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+# FAQ
+
+### What is WebRTC?
+
+WebRTC is an open framework for the web that enables Real Time Communications
+in the browser. It includes the fundamental building blocks for high-quality
+communications on the web, such as network, audio and video components used in
+voice and video chat applications.
+
+These components, when implemented in a browser, can be accessed through a
+JavaScript API, enabling developers to easily implement their own RTC web app.
+
+The WebRTC effort is being standardized on an API level at the W3C and at the
+protocol level at the IETF.
+
+
+### Why should I use WebRTC?
+
+We think you'll want to build your next video-chat style application using
+WebRTC. Here's why:
+
+  * A key factor in the success of the web is that its core technologies --
+    such as HTML, HTTP, and TCP/IP -- are open and freely implementable.
+    Currently, there is no free, high-quality, complete solution available
+    that enables communication in the browser. WebRTC enables this.
+
+  * Already integrated with best-of-breed voice and video engines that have
+    been deployed on millions of endpoints over the last 8+ years. Google does
+    not charge royalties for WebRTC.
+
+  * Includes and abstracts key NAT and firewall traversal technology, using
+    STUN, ICE, TURN, RTP-over-TCP and support for proxies.
+
+  * Builds on the strength of the web browser: WebRTC abstracts signaling by
+    offering a signaling state machine that maps directly to `PeerConnection`.
+    Web developers can therefore choose the protocol of choice for their usage
+    scenario (for example, but not limited to, SIP, XMPP/Jingle, et al.).
+
+
+### What is the Opus audio codec?
+
+[Opus][opus-link] is a royalty-free audio codec defined by IETF
+RFC 6176.  It supports constant and variable bitrate encoding from 6 kbit/s to
+510 kbit/s, frame sizes from 2.5 ms to 60 ms, and various sampling rates from
+8 kHz (with 4 kHz bandwidth) to 48 kHz (with 20 kHz bandwidth, where the
+entire hearing range of the human auditory system can be reproduced).
+
+[opus-link]: http://opus-codec.org/
+
+### What is the iSAC audio codec?
+
+iSAC is a robust, bandwidth-adaptive, wideband and super-wideband voice codec
+developed by Global IP Solutions, and is used in many Voice over IP (VoIP) and
+streaming audio applications. iSAC is used by industry leaders in hundreds of
+millions of VoIP endpoints. This codec is included as part of the WebRTC
+project.
+
+
+### What is the iLBC audio codec?
+
+iLBC is a free narrowband voice codec that was developed by Global IP
+Solutions, and is used in many Voice over IP (VoIP) and streaming audio
+applications. In 2004, the final IETF RFC versions of the iLBC codec
+specification and the iLBC RTP Profile draft became available. This codec is
+included as part of the WebRTC project.
+
+
+### What is the VP8 video codec?
+
+VP8 is a highly-efficient video compression technology developed by the WebM Project. It is the video codec included with WebRTC.
+
+### What is the VP9 video codec?
+
+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].
+
+[vp9-link]: https://developers.google.com/web/updates/2016/01/vp9-webrtc/
+
+### What other components are included in the WebRTC package?
+
+#### Audio
+
+WebRTC offers a complete stack for voice communications. It includes not only
+the necessary codecs, but other components necessary to great user
+experiences. This includes software-based acoustic echo cancellation (AEC),
+automatic gain control (AGC), noise reduction, noise suppression, and
+hardware access and control across multiple platforms.
+
+
+#### Video
+
+The WebRTC project builds on the VP8 codec, introduced in 2010 as part of the
+[WebM Project][webm-link]. It includes components to conceal
+packet loss and clean up noisy images, as well as capture and playback
+capabilities across multiple platforms.
+
+[webm-link]: http://www.webmproject.org/
+
+#### Network
+
+Dynamic jitter buffers and error concealment techniques are included for audio
+and video, which help mitigate the effects of packet loss and unreliable
+networks. Also included are components for establishing a peer-to-peer
+connection using ICE / STUN / Turn / RTP-over-TCP and support for proxies.
+
+
+### How do I access the WebRTC code?
+
+Go to [https://webrtc.googlesource.com/src][webrtc-repo-link].
+
+[webrtc-repo-link]: https://webrtc.googlesource.com/src
+
+
+### How can I test the quality of WebRTC components?
+
+We have put sample applications [here][examples-link].
+
+[examples-link]: https://webrtc.googlesource.com/src/+/main/docs/native-code/development/index.md#example-applications
+
+
+### Are WebRTC components subject to change?
+
+WebRTC is based on a API that is still under development through efforts at
+WHATWG, W3C and IETF. We hope to get to a stable API once a few browser
+vendors have implementations ready for testing. Once the API is stable, our
+goal will be to offer backwards compatibility and interoperability. The WebRTC
+API layer will be our main focus for stability and interoperability. The
+components under it may be modified to improve quality, performance and
+feature set.
+
+
+### WebRTC components are open-source. How do I get the source and contribute code?
+
+Please see [Getting Started][getting-started-link] and
+[Contributing bug fixes][contributing-link] for more information.
+
+[getting-started-link]: https://webrtc.googlesource.com/src/+/main/docs/native-code/development/index.md
+[contributing-link]: https://webrtc.org/support/contributing
+
+
+### To be a Contributor, do I need to sign any agreements?
+
+Yes, each Contributor must sign and return the
+[Contributor License Agreement][cla-link]
+
+[cla-link]: https://developers.google.com/open-source/cla/individual?hl=en
+
+### How can I become a WebRTC committer?
+
+The process of becoming a committer is documented in a
+[separate page][become-a-committer].
+
+[become-a-committer]: https://webrtc.googlesource.com/src/+/refs/heads/main/g3doc/become_a_committer.md
+
+### Do I have to be a programmer to use WebRTC?
+
+Yes, to build WebRTC support into a software application or contribute
+improvements, programming skills are required. However, usage of the
+JavaScript APIs that call WebRTC in the browsers will only require typical web
+development skills.
+
+
+### Is the WebRTC project owned by Google or is it independent?
+
+WebRTC is an open-source project supported by Google, Mozilla and Opera. The
+API and underlying protocols are being developed jointly at the W3C and IETF.
+
+
+### Are the WebRTC components from Google's acquisition of Global IP Solutions?
+
+Yes. Some components, such as VoiceEngine, VideoEngine, NetEQ, AEC, et al. all
+stem from the GIPS acquisition.
+
+
+### What codecs are supported in WebRTC?
+
+The currently supported voice codecs are G.711, G.722, iLBC, and iSAC, and VP8
+is the supported video codec. The list of supported codecs may change in the
+future.
+
+
+### Please explain how WebRTC is free of charge?
+
+Some software frameworks, voice and video codecs require end-users,
+distributors and manufacturers to pay patent royalties to use the intellectual
+property within the software technology and/or codec. Google is not charging
+royalties for WebRTC and its components including the codecs it supports (VP8
+for video and iSAC and iLBC for audio).  For more information, see the [License
+page][license-link].
+
+[license-link]: https://webrtc.googlesource.com/src/+/main/LICENSE
+
+
+### What does this license let me do?
+
+Like most BSD licenses, this license allows you to use the WebRTC code with a
+minimum of restrictions on your use. You can use the code in proprietary
+software as well as open source software.
+
+
+### Do I need to release the source if I make changes?
+
+No, the license does not require you to release source if you make changes.
+However, we would love to see any changes you make and possibly incorporate
+them, so if you want to participate please visit the
+[code review page][code-review-link] and submit some patches.
+
+[code-review-link]: https://webrtc-review.googlesource.com/
+
+
+### Why is there a separate patent grant?
+
+In order to decouple patents from copyright, thus preserving the pure BSD
+nature of the copyright license, the license and the patent grant are
+separate. This means we are using a standard (BSD) open source copyright
+license, and the patent grant can exist on its own. This makes WebRTC
+compatible with all major license scenarios.
+
+
+### What if someone gets the code from Google and gives it to me without changes. Do I have a patent grant from Google?
+
+Yes, you still have the right to redistribute and you still have a patent
+license for Google's patents that cover the code that Google released.
+
+
+### 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?
+
+You still have the right to redistribute but no patent license for the changes
+(if there are any patents covering it). We can't give patent licenses for
+changes people make after we distribute the code, as we have no way to predict
+what those changes will be. Other common licenses take the same approach,
+including the Apache license.
+
+
+### 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?
+
+Yes, you still have the right to redistribute and you still have a patent
+license for Google's patents that cover the code that Google released.
+
+
+### What if my competitor uses the code and brings patent litigation against me for something unrelated to the code. Do they still have a patent license?
+
+Yes, they still have the right to redistribute and they still have a patent
+license for Google's patents that cover the code that Google released.