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No-Try: True Bug: None Change-Id: I6737b5adee6e3db45b5825a01460b66690eec896 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168883 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30571}
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docs/faq.md
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# What is WebRTC?
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WebRTC is an open framework for the web that enables Real Time Communications
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in the browser. It includes the fundamental building blocks for high-quality
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communications on the web, such as network, audio and video components used in
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voice and video chat applications.
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These components, when implemented in a browser, can be accessed through a
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JavaScript API, enabling developers to easily implement their own RTC web app.
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The WebRTC effort is being standardized on an API level at the W3C and at the
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protocol level at the IETF.
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# Why should I use WebRTC?
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We think you'll want to build your next video-chat style application using
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WebRTC. Here's why:
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* A key factor in the success of the web is that its core technologies --
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such as HTML, HTTP, and TCP/IP -- are open and freely implementable.
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Currently, there is no free, high-quality, complete solution available
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that enables communication in the browser. WebRTC enables this.
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* Already integrated with best-of-breed voice and video engines that have
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been deployed on millions of endpoints over the last 8+ years. Google does
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not charge royalties for WebRTC.
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* Includes and abstracts key NAT and firewall traversal technology, using
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STUN, ICE, TURN, RTP-over-TCP and support for proxies.
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* Builds on the strength of the web browser: WebRTC abstracts signaling by
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offering a signaling state machine that maps directly to `PeerConnection`.
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Web developers can therefore choose the protocol of choice for their usage
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scenario (for example, but not limited to, SIP, XMPP/Jingle, et al.).
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# What is the Opus audio codec?
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[Opus][opus-link] is a royalty-free audio codec defined by IETF
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RFC 6176. It supports constant and variable bitrate encoding from 6 kbit/s to
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510 kbit/s, frame sizes from 2.5 ms to 60 ms, and various sampling rates from
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8 kHz (with 4 kHz bandwidth) to 48 kHz (with 20 kHz bandwidth, where the
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entire hearing range of the human auditory system can be reproduced).
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[opus-link]: http://opus-codec.org/
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# What is the iSAC audio codec?
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iSAC is a robust, bandwidth-adaptive, wideband and super-wideband voice codec
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developed by Global IP Solutions, and is used in many Voice over IP (VoIP) and
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streaming audio applications. iSAC is used by industry leaders in hundreds of
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millions of VoIP endpoints. This codec is included as part of the WebRTC
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project.
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# What is the iLBC audio codec?
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iLBC is a free narrowband voice codec that was developed by Global IP
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Solutions, and is used in many Voice over IP (VoIP) and streaming audio
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applications. In 2004, the final IETF RFC versions of the iLBC codec
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specification and the iLBC RTP Profile draft became available. This codec is
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included as part of the WebRTC project.
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# What is the VP8 video codec?
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VP8 is a highly-efficient video compression technology developed by the WebM Project. It is the video codec included with WebRTC.
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# What is the VP9 video codec?
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Similar to VP8, VP9 is also from the WebM Project. Its a next-generation open video codec. From Chrome 48 on desktop and Android, VP9 will be an optional video codec for video calls. More details in [Google Developers][vp9-link].
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[vp9-link]: https://developers.google.com/web/updates/2016/01/vp9-webrtc/
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# What other components are included in the WebRTC package?
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## Audio
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WebRTC offers a complete stack for voice communications. It includes not only
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the necessary codecs, but other components necessary to great user
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experiences. This includes software-based acoustic echo cancellation (AEC),
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automatic gain control (AGC), noise reduction, noise suppression, and
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hardware access and control across multiple platforms.
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## Video
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The WebRTC project builds on the VP8 codec, introduced in 2010 as part of the
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[WebM Project][webm-link]. It includes components to conceal
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packet loss and clean up noisy images, as well as capture and playback
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capabilities across multiple platforms.
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[webm-link]: http://www.webmproject.org/
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## Network
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Dynamic jitter buffers and error concealment techniques are included for audio
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and video, which help mitigate the effects of packet loss and unreliable
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networks. Also included are components for establishing a peer-to-peer
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connection using ICE / STUN / Turn / RTP-over-TCP and support for proxies.
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# How do I access the WebRTC code?
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Go to [https://webrtc.googlesource.com/src][webrtc-repo-link]
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[webrtc-repo-link]: https://webrtc.googlesource.com/src
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# How can I test the quality of WebRTC components?
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We have put sample applications [here][examples-link].
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[examples-link]: https://webrtc.googlesource.com/src/+/refs/heads/master/docs/native-code/development/index.md#example-applications
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# Are WebRTC components subject to change?
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WebRTC is based on a API that is still under development through efforts at
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WHATWG, W3C and IETF. We hope to get to a stable API once a few browser
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vendors have implementations ready for testing. Once the API is stable, our
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goal will be to offer backwards compatibility and interoperability. The WebRTC
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API layer will be our main focus for stability and interoperability. The
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components under it may be modified to improve quality, performance and
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feature set.
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# WebRTC components are open-source. How do I get the source and contribute code?
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Please see [Getting Started][getting-started-link] and
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[Contributing bug fixes][contributing-link] for more information.
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[getting-started-link]: https://webrtc.googlesource.com/src/+/refs/heads/master/docs/native-code/development/index.md
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[contributing-link]: https://webrtc.org/support/contributing
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# To be a Contributor, do I need to sign any agreements?
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Yes, each Contributor must sign and return the
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[Contributor License Agreement][cla-link]
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[cla-link]: https://developers.google.com/open-source/cla/individual?hl=en
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# Do I have to be a programmer to use WebRTC?
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Yes, to build WebRTC support into a software application or contribute
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improvements, programming skills are required. However, usage of the
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JavaScript APIs that call WebRTC in the browsers will only require typical web
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development skills.
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# Is the WebRTC project owned by Google or is it independent?
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WebRTC is an open-source project supported by Google, Mozilla and Opera. The
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API and underlying protocols are being developed jointly at the W3C and IETF.
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# Are the WebRTC components from Google's acquisition of Global IP Solutions?
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Yes. Some components, such as VoiceEngine, VideoEngine, NetEQ, AEC, et al. all
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stem from the GIPS acquisition.
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# What codecs are supported in WebRTC?
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The currently supported voice codecs are G.711, G.722, iLBC, and iSAC, and VP8
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is the supported video codec. The list of supported codecs may change in the
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future.
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# Please explain how WebRTC is free of charge?
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Some software frameworks, voice and video codecs require end-users,
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distributors and manufacturers to pay patent royalties to use the intellectual
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property within the software technology and/or codec. Google is not charging
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royalties for WebRTC and its components including the codecs it supports (VP8
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for video and iSAC and iLBC for audio). For more information, see the [License
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page][license-link].
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[license-link]: https://webrtc.googlesource.com/src/+/refs/heads/master/docs/license/index.md
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# What does this license let me do?
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Like most BSD licenses, this license allows you to use the WebRTC code with a
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minimum of restrictions on your use. You can use the code in proprietary
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software as well as open source software.
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# Do I need to release the source if I make changes?
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No, the license does not require you to release source if you make changes.
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However, we would love to see any changes you make and possibly incorporate
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them, so if you want to participate please visit the
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[code review page][code-review-link] and submit some patches.
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[code-review-link]: https://webrtc-review.googlesource.com/
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# Why is there a separate patent grant?
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In order to decouple patents from copyright, thus preserving the pure BSD
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nature of the copyright license, the license and the patent grant are
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separate. This means we are using a standard (BSD) open source copyright
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license, and the patent grant can exist on its own. This makes WebRTC
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compatible with all major license scenarios.
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# What if someone gets the code from Google and gives it to me without changes. Do I have a patent grant from Google?
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Yes, you still have the right to redistribute and you still have a patent
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license for Google's patents that cover the code that Google released.
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# What if someone makes a change to the code and gives it to me. Do I have a patent license from Google for that change?
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You still have the right to redistribute but no patent license for the changes
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(if there are any patents covering it). We can't give patent licenses for
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changes people make after we distribute the code, as we have no way to predict
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what those changes will be. Other common licenses take the same approach,
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including the Apache license.
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# What if Google receives or buys a patent that covers the code I receive sometime after I receive the code. Do I have a patent grant for that patent?
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Yes, you still have the right to redistribute and you still have a patent
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license for Google's patents that cover the code that Google released.
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# What if my competitor uses the code and brings patent litigation against me for something unrelated to the code. Does he or she still have a patent license?
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Yes, he/she still has the right to redistribute and he/she still has a patent
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license for Google's patents that cover the code that Google released.
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