• Webrtc Firefox Extension
• Webrtc Control Firefox Chrome
• Webrtc Control Firefox
• Webrtc Control Firefox Web

If WebRTC is enabled on Firefox because you have been using it and you plan to use it again in the future, installing a plugin makes more sense since switching it on and off will take a long time. Here are the steps: Open the Add-ons page. In the search box, type in WebRTC Control. The browsers that run WebRTC are all open source (e.g., WebKit for Safari, Chromium for Chrome & Edge, Gecko for Firefox, etc.) and thoroughly tested. The core WebRTC project is also open source and accompanied by dozens of different implementations actively supported by thousands of developers who are constantly reviewing and improving WebRTC. WebRTC app and indicates the version of Meeting Server where this feature / fix is available. L Any upcoming changes in browsers affecting the app, affected versions with recommended workarounds. WebRTC is still an evolving technology and frequent changes are done by browser vendors.

WebRTC is a new front in the long war for an open and unencumbered web. Brendan Eich, inventor of JavaScript

Real-time communication without plugins

Imagine a world where your phone, TV, and computer could communicate on a common platform. Imagine it was easy to add video chat and peer-to-peer data sharing to your web app. That's the vision of WebRTC.

• In the simplest of terms, WebRTC enables multimedia communications (video and audio chats) between two or more people in their browsers, without having to download any additional software. All major browsers, including Google Chrome, Firefox, Microsoft Edge, Apple Safari, and Opera, support this technology.
• On your Firefox browser, go to the menu, which is at the top right shown with three vertical lines. Now, click the option of “Add-on”. On the left side of your screen, press “Search”. After this, enter the name of the add-on “WebRTC control” and click Install.

Want to try it out? WebRTC is available on desktop and mobile in Google Chrome, Safari, Firefox, and Opera. A good place to start is the simple video chat app at appr.tc:

1. Open appr.tc in your browser.
2. Click Join to join a chat room and let the app use your webcam.
3. Open the URL displayed at the end of the page in a new tab or, better still, on a different computer.

Quick start

Haven't got time to read this article or only want code?

Webrtc Firefox Extension

1. To get an overview of WebRTC, watch the following Google I/O video or view these slides:

2. If you haven't used the getUserMedia API, see Capture audio and video in HTML5 and simpl.info getUserMedia.
3. To learn about the RTCPeerConnection API, see the following example and simpl.info RTCPeerConnection.
4. To learn how WebRTC uses servers for signaling, and firewall and NAT traversal, see the code and console logs from appr.tc.
5. Can’t wait and just want to try WebRTC right now? Try some of the more-than 20 demos that exercise the WebRTC JavaScript APIs.
6. Having trouble with your machine and WebRTC? Visit the WebRTC Troubleshooter.

Alternatively, jump straight into the WebRTC codelab, a step-by-step guide that explains how to build a complete video chat app, including a simple signaling server.

A very short history of WebRTC

One of the last major challenges for the web is to enable human communication through voice and video: real-time communication or RTC for short. RTC should be as natural in a web app as entering text in a text input. Without it, you're limited in your ability to innovate and develop new ways for people to interact.

Historically, RTC has been corporate and complex, requiring expensive audio and video technologies to be licensed or developed in house. Integrating RTC technology with existing content, data, and services has been difficult and time-consuming, particularly on the web.

Gmail video chat became popular in 2008 and, in 2011, Google introduced Hangouts, which uses Talk (as did Gmail). Google bought GIPS, a company that developed many components required for RTC, such as codecs and echo cancellation techniques. Google open sourced the technologies developed by GIPS and engaged with relevant standards bodies at the Internet Engineering Task Force (IETF) and World Wide Web Consortium (W3C) to ensure industry consensus. In May 2011, Ericsson built the first implementation of WebRTC.

WebRTC implemented open standards for real-time, plugin-free video, audio, and data communication. The need was real:

• Many web services used RTC, but needed downloads, native apps, or plugins. These included Skype, Facebook, and Hangouts.
• Downloading, installing, and updating plugins is complex, error prone, and annoying.
• Plugins are difficult to deploy, debug, troubleshoot, test, and maintain—and may require licensing and integration with complex, expensive technology. It's often difficult to persuade people to install plugins in the first place!

The guiding principles of the WebRTC project are that its APIs should be open source, free, standardized, built into web browsers, and more efficient than existing technologies.

Where are we now?

WebRTC is used in various apps, such as Google Meet. WebRTC has also been integrated with WebKitGTK+ and Qt native apps.

WebRTC implements these three APIs:

• MediaStream (also known as getUserMedia)

The APIs are defined in these two specs:

All three APIs are supported on mobile and desktop by Chrome, Safari, Firefox, Edge, and Opera.

getUserMedia: For demos and code, see WebRTC samples or try Chris Wilson's amazing examples that use getUserMedia as input for web audio.

RTCPeerConnection: For a simple demo and a fully functional video-chat app, see WebRTC samples Peer connection and appr.tc, respectively. This app uses adapter.js, a JavaScript shim maintained by Google with help from the WebRTC community, to abstract away browser differences and spec changes.

RTCDataChannel: To see this in action, see WebRTC samples to check out one of the

Developer tools

• WebRTC stats for an ongoing session can be found at:
  • chrome://webrtc-internals in Chrome
  • opera://webrtc-internals in Opera
  • about:webrtc in Firefox
• Cross browser interop notes
• adapter.js is a JavaScript shim for WebRTC maintained by Google with help from the WebRTC community that abstracts vendor prefixes, browser differences, and spec changes.
• To learn more about WebRTC signaling processes, check the appr.tc log output to the console.
• If it's all too much, you may prefer to use a WebRTC framework or even a complete WebRTC service.
• Bug reports and feature requests are always appreciated:

Learn more

• Alan B. Johnston and Daniel C. Burnett maintain a WebRTC book now in its third edition in print and eBook formats at webrtcbook.com.
• webrtc.org is home to all things WebRTC, including demos, documentation, and discussion.
• discuss-webrtc is a Google Group for technical WebRTC discussion.
• Google Developers Talk documentation provides more information about NAT traversal, STUN, relay servers, and candidate gathering.
• Stack Overflow is a good place to look for answers and ask questions about WebRTC.

Standards and protocols

• W3C Editor's Draft: Media Capture and Streams (also known as getUserMedia)
• IETF RTCWEB Working Group Internet-Draft: Web Real-Time Communication Use-cases and Requirements

WebRTC support summary

MediaStream and getUserMedia APIs

• Chrome desktop 18.0.1008 and higher; Chrome for Android 29 and higher
• Opera 18 and higher; Opera for Android 20 and higher
• Opera 12, Opera Mobile 12 (based on the Presto engine)
• Firefox 17 and higher
• Microsoft Edge 16 and higher
• Safari 11.2 and higher on iOS, and 11.1 and higher on MacOS
• UC 11.8 and higher on Android
• Samsung Internet 4 and higher

RTCPeerConnection API

• Chrome desktop 20 and higher; Chrome for Android 29 and higher (flagless)
• Opera 18 and higher (on by default); Opera for Android 20 and higher (on by default)
• Firefox 22 and higher (on by default)
• Microsoft Edge 16 and higher
• Safari 11.2 and higher on iOS, and 11.1 and higher on MacOS
• Samsung Internet 4 and higher

RTCDataChannel API

• Experimental version in Chrome 25, but more stable (and with Firefox interoperability) in Chrome 26 and higher; Chrome for Android 29 and higher
• Stable version (and with Firefox interoperability) in Opera 18 and higher; Opera for Android 20 and higher
• Firefox 22 and higher (on by default)

For more detailed information about cross-platform support for APIs, such as getUserMedia and RTCPeerConnection, see caniuse.com and Chrome Platform Status.

Native APIs for RTCPeerConnection are also available at documentation on webrtc.org.

The good news is that WebRTC is very secure. The bad news is that no system is without flaws and WebRTC security is complicated. In this post, I will discuss how WebRTC approaches security, identify some known weak areas, and discuss how WebRTC is continuing to improve its security.

Multiple Security Approaches

WebRTC is a complex, layered protocol that exists in a complex layered ecosystem of application code, browsers, native devices, and infrastructure elements. WebRTC approaches security from several different angles. First, it is secured at the protocol level. Second, using the browser as a reference, it requires a protected and private execution environment. Third, it follows best security practices by engaging its community of developers.

Protocol Layer — Mandatory Encryption Specifications

Mandatory Media Encryption

Unlike other VoIP and video conferencing technologies, encryption is mandated in WebRTC. To send video, voice, or data between two peers in WebRTC, the information must be encrypted with Secure Real Time Protocol (SRTP). SRTP encrypts the session, so no one can decode the message without the proper encryption keys. In fact, the unencrypted version of RTP is explicitly forbidden by the IETF specifications that define WebRTC.

Mandatory Secure Encryption Key Exchange

In addition, the WebRTC specifications mandate secure setup of the encryption channel to make it difficult to obtain the encryption keys. Locking your house doesn’t do much good if it is easy to find the key under your doormat. Many key exchange mechanisms such as SDES, MIKEY, and ZRTP may be used to set up this encrypted channel. Systems like SDES and MIKEY leverage the signaling channel to transmit this key data. This means if the signaling channel is compromised, the data could be unencrypted by a third party. To prevent this possibility, the WebRTC specifications mandate the use of DTLS-SRTP where keys are exchanged directly between peers on the media plane. Despite the fact that SDES is still widely used in many VoIP systems, it is specifically barred from use in WebRTC because it is not secure enough.

Secure Signaling

Lastly, WebRTC requires a secure connection between the web server that handles signaling and the peer client. This helps to keep the information in that signaling channel secure and makes it more difficult for an attacker to act as a man-in-the-middle and quietly take over the session. Signaling is secured using the HTTPS protocol — the same one most websites now use.

WebRTC security points in a peer-to-peer architecture providing end-to-end encryption

In live streaming environments, the server acts as both a signaling server and WebRTC media peer, but the same secure interfaces are used.

WebRTC security points in a live streaming architecture like Wowza Streaming Engine

WebRTC Is Secured by the Browser

WebRTC is further secured by operating in a browser sandbox. Web browsers are the most commonly used applications and have developed sophisticated security and privacy features. These features help to isolate web applications, keep sensitive user information like credit cards secure, and project hijacking of the browser to launch attacks.

Browser Security and Privacy Protections

Browser vendors are subject to strict security standards as defined by the W3C and underlying Internet specifications, like those for the WebRTC protocol. This is one top of security mechanisms at the Operating System level like limiting which applications can use the camera. Competition among major browsers like Chrome, Firefox, Edge, and Safari among others has also made them very sensitive to their users security and privacy needs in general, and particularly for WebRTC. Specific examples of WebRTC security and privacy controls include:

1. HTTPS: Use of HTTPS is required to access WebRTC features (with some small exceptions for development)
2. Media access permissions: Users must explicitly grant permissions to individual sites before accessing camera, microphone, or screen sharing video
3. Visual usage indicators: Prominent indicators must show when your camera, microphone, and screen sharing is being used.
4. Anonymizing device information: Device information remains hidden until the user has given some permissions for the site.
5. IP leakage protections: Limitations and options on sharing IP address information help avoid privacy and tracking issues.

Implementation of these features does vary, but most are similar across major browsers.

Mobile OSs Have Similar Controls

Many native mobile applications also incorporate an embedded browser framework to utilize some or all of these features. Even if not using an embedded browser, major mobile operating systems like Android and iOS implement similar controls and have additional security and privacy checks through their app store submission procedures.

WebRTC Is Secured by the Community

There are 2 main philosophies to security:

1. Security by obscurity: Keep the mechanisms of your system secret to make it harder to discover and compromise.
2. Security by design: Make the mechanisms of your system open, invite others to try to break in, and improve the design through feedback.

Webrtc Control Firefox Chrome

Security researchers and organizations generally do not condone the first philosophy. For example, the National Institute of Standards and Technology (NIST)says “System security should not depend on the secrecy of the implementation or its components.” It is well known that secrecy never lasts forever, especially in the face of motivated adversaries.

WebRTC subscribes to security by design philosophy. Communications across browsers and devices over the Internet requires open standards, so WebRTC has little choice at the protocol level. However, for security by design to work, it needs to have broad scrutiny. The browsers that run WebRTC are all open source (e.g., WebKit for Safari, Chromium for Chrome & Edge, Gecko for Firefox, etc.) and thoroughly tested. The core WebRTC project is also open source and accompanied by dozens of different implementations actively supported by thousands of developers who are constantly reviewing and improving WebRTC in all aspects, including security.

Security Depends on the Service

WebRTC mandates encryption at the protocol level. It is subject to rigorous privacy and security controls when run inside a browser environment, but what about services that don’t use the browser? While WebRTC is primarily designed for browser-to-browser communication, many different infrastructure devices are commonly used in WebRTC services.

The table below summarizes common types of WebRTC server infrastructure and high-level security implications:

Inherent Internet Security Issues

Signaling services are essentially web application servers and need to be secured like any application. Users should be weary of who they are connecting too. Fortunately, browser and app store safeguards minimize bad providers – but bad users are another problem. “Zoom bombing” is an example where someone obtains video conferencing meeting information to join and disrupt the meeting. WebRTC service providers can prevent this by giving their users authentication mechanisms that restrict entry to authorized users and leverage moderation controls to remove and block bad actors quickly. For example, Wowza has APIs that control WebRTC stream access and duration.

Media Server Risks

While other servers like STUN and TURN are sometimes needed, these never gain access to unencrypted media, and thus don’t present much of a risk. Other servers, particularly media servers like Selective Forwarding Units (SFUs) that enable multi-party video conference calls or live streaming servers, introduce more difficult risks. These servers decrypt the media before restransmitting it with new encryption keys. Oftentimes this decryption cannot be avoided. If the media server needs to manipulate the media then it must perform decryption to access that media. For example, in live streaming networks, the media server needs to decrypt the media to resize, recompose, and convert the format for use by a Content Delivery Network (CDN).

If these servers are compromised, then the user media streams could be at risk. It is important that media server operators follow best security practices when deploying their infrastructure and managing people to prevent unauthorized access. Much like browsers do for end users, these servers should avoid unintentional caching of sensitive unencrypted data. Sensitive information should never be at rest where an attacker could access it. The servers should also isolate the media streams internally from from other processes that could nefariously access it.

The level of security will also need to vary based on the application. For example, recordings may be needed for archiving and intentional redistribution. In that case the stream will certainly need to be saved to disk and made available, but that should be done in a controller, secure manner.

Is WebRTC Secure Enough?

No software system is perfectly secure, and WebRTC is no exception. For example, a Google Project Zero security researcher recently published a major exploit that worked on 7 out 14 of the most popular Android WebRTC applications. One one hand, it is very bad that such a serious issue could have made its way into applications used on billions of devices. On the other hand, the fact that there is deep security vulnerability research done in the public domain and that all but one of these applications rapidly fixed their issues is encouraging.

Like most software, there are a few general rules WebRTC developers can follow to minimize their attack surface for vulnerabilities:

• Keep core WebRTC libraries up to date. Old code generally has more vulnerabilities.
• Pay attention to bugs and security notices. Major WebRTC projects and browsers are generally very proactive in their notifications (if you know where to look).
• Turn off pieces of code you don’t use. WebRTC is a large and comprehensive system that many apps only need a part off; minimize the attack surface.
• Test and conduct RTC-specific security research. If you don’t find your issues, someone else will eventually.
• Secure your infrastructure – WebRTC may be secure, but if your web or media servers are insecure it could compromise the system. For example, features like Wowza’s publication authentication features to limit the attack opportunity for stream hijacking.

WebRTC is used every day by billions of people. Its security is certainly not perfect, but it provides the most secure approach by mandating security at a low level, working with an established security sandbox in many cases (the browser), and encouraging review by a large and very active community.

Webrtc Control Firefox

Additional Resources:

Webrtc Control Firefox Web

About Chad Hart