Webrtc extension. Supported desktop browsers

WebRTC (short for Web real-time communications) is a technology that allows audio and video streaming between browsers and mobile applications.

The development of this technology competes with Skype. WebRTC can be used to organize video conferencing directly in the browser. The project is open source and is actively promoted by Google and in particular by the Google Chrome browser development team.

Thanks to WebRTC technology, user browsers can transfer data to each other directly. WebRTC does not need a separate server to store and process data. All data is processed directly by browsers and mobile applications of end users.

WebRTC technology is supported by all popular browsers Mozilla Firefox, Opera, Google Chrome (and all browsers based on Google Chrome), as well as mobile applications based on Android and iOS.

Danger of WebRTC

The danger of WebRTC technology lies in determining your real IP address. Since the connection is direct with another user, browser, website or mobile application, the network settings are ignored. To create audio and video links, browsers must exchange external and local IP addresses.

Anonymous VPN service solves this problem and hides the real IP address. The maximum that can be detected is the local IP address assigned to the user by the VPN network. This is not dangerous, as the same local IP addresses will be shown if you are using a router to distribute the Internet.

If you are using a proxy, then WebRTC will be able to determine your real IP address behind the proxy, or the IP address of the VPN server if you are using the VPN + proxy chain.

WebRTC also detects your real IP address when using the Tor network.

The best solution is to disable WebRTC technology if you are not using it.

How to disable WebRTC in browsers

Quick navigation on this page.

How to disable WebRTC in Mozilla Firefox

Mozilla Firefox is the only browser that allows you to disable WebRTC technology without installing additional plugins.

Manual setting

If you are not using WebRTC technology, you can completely disable it. In the case when it is necessary to use WebRTC from time to time it is more convenient.

To disable WebRTC technology in Mozilla Firefox, enter the following text in the browser address bar and press the Enter button.

About: config

Click the I accept the risk button.

Do the following:

1. Enter text into the search box and press Enter.
media.peerconnection.enabled
2. Right-click on the line and select Switch. Or double click on the line.

After these actions, WebRTC will be disabled.

Configuration via WebRTC Control plugin

If you use WebRTC technology, then disabling and enabling via settings will take a long time. Install a plugin that will help you enable and disable WebRTC in 1 click.

Open Add-ons.

Select:

1. Search section
2. Enter the plugin name in the search box: WebRTC Control
3. Click the Install button

How to disable WebRTC in Opera browser

To disable WebRTC in the Opera browser, go to the Extensions gallery.

Follow these steps:

1. Enter the plugin name in the search bar: WebRTC Control
2. Click on the plugin

Click Add to Opera.

Activate the plugin. The plugin icon should turn blue to block WebRTC.

How to disable WebRTC in Google Chrome

To disable WebRTC in the Google Chrome browser, go to the Extensions section.

Scroll down the page and click More extensions.

Follow these steps:

1. Enter the plugin name in the search box: WebRTC Control
2. Click the Install button.

Activate the plugin. The plugin icon should turn blue to block WebRTC.

How to disable WebRTC in Yandex Browser

To disable WebRTC in Yandex Browser, go to the Add-ons section.

Scroll down the page and click Yandex Browser extensions catalog.

Follow the steps:

1. Enter the plugin name in the search bar: WebRTC Control
2. Click on the plugin to install.

Click Add to Yandex Browser.

Click Install Extension.

Activate the plugin. The plugin icon should turn blue to block WebRTC.

TrueConf Server users can connect to video conferencing through a browser using WebRTC technology.

Popular browsers have started to include support for Web Real-Time Communications (WebRTC) in their builds since 2012. Google Chrome became the pioneer, after which the rest of the browsers took up the baton.

Today, connection using WebRTC technology is possible in the following browsers:

Google Chrome *		Mozilla FireFox						Microsoft

* - and all browsers based on the Chromium engine

WebRTC Connection: Supported Browsers

Google Сhrome

In our list, the Google Chrome browser took the honorable first place due to the fact that it was he who pioneered and was the first to support the WebRTC technology in 2012, starting with version 17. Before that, no browser could make video calls without installing special client applications. Therefore, the very idea of ​​creating the ability to capture audio and video streams and their subsequent playback directly in the browser was accepted by users with a bang.

However, it should be noted that browser calls existed long before the advent of WebRTC, let us recall at least the notorious Flash and Java. However, Java's problem is the complexity of the implementation of DSP libraries (no echo cancellation, audio delay), and Flash must be downloaded and installed on your PC, which, of course, takes time. These problems do not exist in WebRTС.

By the way, we have already managed to compare the new trend - WebRTC with Flash technology. You can see their "fight" in our blog.

With each new version, Google Chrome supported all the new features of WebRTC, and by now we see the result - a full-fledged browser client for video calls that does not require the installation of additional extensions.

Mozilla Firefox

The Mozilla Firefox browser has supported WebRTC since its 18th Aurora build, which is
initially required manual configuration of the video calls functionality. By default, WebRTC did not function, so before giving permission to request site access to the microphone and camera, users were prompted to visit the advanced settings page ( about: config). However, users did not suffer for long - already in 2013 Firefox released version 22, which included WebRTC support by default.

Opera

In Opera, just like in the previous browser, support for the WebRTC standard was implemented in version 18, which was released in December 2013. It was in this assembly that the interface support was enabled by default. getUserMedia API, which is an integral component of the WebRTC technology. Using this interface, the Opera web browser could access the webcam and microphone of the user's computer.

Internet Explorer

In Internet Explorer, a technology similar to WebRTC exists under a fundamentally different name - ORTC. This is a new free project, which, according to Microsoft, will become an excellent alternative to WebRTC, and maybe even completely re-qualify in WebRTC 1.1. The standard has supported the best proven audio codecs - Opus, G.711 and G.722, and of course has implemented H.264 for video encoding.

Microsoft Edge

Support for WebRTC technology in Microsoft Edge was implemented for Windows 10 in Build 15019, which was released in January 2017. It was in this assembly that WebRTC 1.0 was enabled by default. Prior to this, Microsoft Edge supported ORTC starting with EdgeHTML 13 for Windows 10 build 15011. It is now possible to exchange messages in real time in Microsoft Edge using WebRTC technologies.

Safari

Connecting via WebRTC to a conference on TrueConf Server is available only in the desktop version of the Safari browser (11+).

Apple released updates for its Safari browser in September 2017. The updated version 11 of Safari for desktop has full support for WebRTC. Older versions of the Safari browser used Flash Player or WebRTC plugins as a fallback mechanism.

WebRTC on mobile browsers

By the way, in March 2014, the 20th build of Opera was released with support for Web Real-Time Communication for Android mobile devices.

Other browsers for Android supported WebRTC a little earlier: Google Chrome Beta 29 with support for the standard was released in July 2013, a couple of months later - in September of the same year - Mozilla Firefox for Android released build 24 with the ability to make video calls from the browser.

In October 2014 Ericsson's mobile browser was released for iOS devices. A web browser called Bowser and the OpenWebRTC platform pioneered the WebRTC standard for Apple devices.

A detailed comparison of the nuances of WebRTC technology support in browsers is given.

WebRTC in TrueConf solutions

When participating in a TrueConf conference via WebRTC, you can:

• enable / disable the camera, sound and microphone;
• exchange text messages;
• use the built-in conference manager with a list of participants;
• view presentations broadcast through client applications;
• maximize the video window to full screen;
• broadcast a picture of your screen to other participants.

To connect to a conference created on the TrueConf server, you need to follow a special link that is distributed by the administrator to all users. When you go to the browser, you must provide access to your microphone and camera so that the rest of the conference participants can see and hear you.

In the Mozilla Firefox browser, the microphone access request looks like this:

By allowing access to your devices, you will be taken to the conference:

You can read more about the process of creating a conference in or in, and you can test the listed features using TrueConf Server Free.

Hello. Tell me how to properly disable WebRTC in Firefox and other browsers?

Disabling WebRTC is not difficult, but before we move on to the instructions for disabling, let's figure out what kind of beast it is and why a regular user needs to disable it at all.

What is WebRTC?

WebRTC (Web Real-Time Communication) is an open source standard that allows real-time streaming of video and audio data through a browser using P2P technology.

This technology is built-in and enabled by default in all modern browsers: Mozilla Firefox, Google Chrome, Chromium and Opera.

Why is WebRTC dangerous?

Why is the enabled WebRTC feature dangerous? Because P2P, during the exchange of information, asks for the user's real IP address, and WebRTC kindly provides this information to everyone. Even if the user works through this miracle of technology, an attacker can easily find out the user's IP address. And using vulnerabilities can, of course, your real IP.

How to check if WebRTC is enabled?

To check your browser, you need to go to these sites:

The next screen says that the WebRTC function is enabled in your browser. In addition, the site provides other interesting information.

How do I disable WebRTC?

Of all modern browsers, Firefox is the best trainable. And in this particular case, Firefox showed its best side, enabling the user to disable WebRTC using hidden settings without using third-party add-ons.

Disable WebRTC Firefox

To disable in the Firefox browser, enter the about: config command in the address bar, after which this message will appear.

We press the button "I promise ..." and continue on.

In the settings window, in the search bar (not in the address bar!), As shown in the screenshot below, enter the command media.peerconnection.enabled. The line we need will appear. Click on it with the right mouse click and select the first item "Toggle" in the drop-down menu.

After switching to the "Value" field, you should see the "False" parameter. Now we close this window and reload the browser.

Another way is to install a special extension. But I prefer and advise you to perform this operation yourself. I do not like installing programs on a computer, especially in a browser.

There is an even easier way - download the ConfigFox utility, which, in addition to this operation, can significantly improve the privacy and anonymity of the Firefox browser. We wrote about this program in the article ““. I highly recommend using this utility to all users of the Mozilla Firefox browser. The program does not install itself into the browser, but simply allows you to change the settings file.

Disable WebRTC Chrome

In the Google Chrome browser, things are a little more complicated. In Chrome, there is no way to disable this feature in the browser itself. To do this, you need to download a special add-on, it is called WebRTC Block. You can download and install the add-on via the direct link. We have not tested this extension and we can not give any guarantees.

There is also an add-on that can also help solve this problem. In my opinion this is the best way to solve the WebRTC problem in Chrome.

It will be more difficult for an inexperienced user with this extension, but if you are an advanced user, I strongly advise you to dig deeper into it.

If you know of other ways to solve this problem in the Chrome browser, write in the comments.

Disable WebRTC Opera / Yandex Browser

There are several plugins for the Opera browser: and. I have not personally tested it, try and write what helped and what did not.

In conclusion, I want to say that at the moment there is no reliable one hundred percent way to disable WebRTC in Chromium browsers such as: Chrome, Yandex, Opera, etc. Therefore, I advise everyone who uses a VPN and who cares about anonymity to temporarily stop using these browsers. I think in the near future, this hole will be closed and you can return to them. In the meantime, you can temporarily move to Firefox.

That's all. In the next articles we will talk about the anonymity and reliability of VPN and public proxies. It will be fun, we will break stereotypes. You will love it;)!

WebRTC (Web Real Time Communications) is a standard that describes the transfer of streaming audio data, video data and content from the browser and to the browser in real time without installing plugins or other extensions. The standard allows you to turn your browser into an endpoint video conferencing terminal, you just need to open a web page to start chatting.

What is WebRTC?

In this article, we'll go over everything there is to know about WebRTC technology for the average user. Let's consider the advantages and disadvantages of the project, reveal some secrets, tell you how it works, where and what WebRTC is used for.

What you need to know about WebRTC?

Evolution of video communication standards and technologies

How WebRTC works

Client side

• User opens a page containing HTML5 tag
• The browser requests access to the user's webcam and microphone.
• JavaScript code on the user page controls connection parameters (IP addresses and ports of the WebRTC server or other WebRTC clients) for NAT and Firewall traversal.
• When receiving information about the interlocutor or about the stream with the conference mixed on the server, the browser starts negotiating the audio and video codecs used.
• The encoding process begins and the transfer of streaming data between WebRTC clients (in our case, between the browser and the server).

On the WebRTC server side

To exchange data between two participants, a video server is not required, but if you need to combine several participants in one conference, a server is required.

The video server will receive media traffic from various sources, transform it and send it to users who use WebRTC as a terminal.

Also, the WebRTC server will receive media traffic from WebRTC peers and transmit it to conference participants who use applications for desktop computers or mobile devices, if any.

Advantages of the standard

• No software installation required.
• Very high quality of communication, thanks to:
  • Using modern video (VP8, H.264) and audio codecs (Opus).
  • Automatic adjustment of the stream quality to the connection conditions.
  • Built-in echo and noise cancellation system.
  • Automatic adjustment of the participant microphone sensitivity level (AGC).
• High level of security: all connections are protected and encrypted according to the TLS and SRTP protocols.
• There is a built-in mechanism for capturing content, such as the desktop.
• The ability to implement any control interface based on HTML5 and JavaScript.
• The ability to integrate the interface with any back-end systems using WebSockets.
• Open source project - can be embedded in your product or service.
• True cross-platform: the same WebRTC application will work equally well on any operating system, desktop or mobile, as long as the browser supports WebRTC. This significantly saves resources for software development.

Disadvantages of the standard

• To organize group audio and video conferencing, a video conferencing server is required, which would mix video and sound from the participants, since the browser cannot synchronize several incoming streams with each other.
• All WebRTC solutions are incompatible with each other. the standard describes only the methods of transmitting video and sound, leaving the implementation of methods for addressing subscribers, tracking their availability, exchanging messages and files, scheduling and other things for the vendor.
• In other words, you cannot call from one developer's WebRTC application to another developer's WebRTC application.
• Mixing group conferences requires a lot of computing resources, therefore this type of video communication requires the purchase of a paid subscription or investment in its infrastructure, where each conference requires 1 physical core of a modern processor.

WebRTC Secrets: How Vendors Benefit From Disruptive Web Technology

WebRTC for the video conferencing market

Increase in the number of video conferencing terminals

WebRTC technology has had a strong impact on the development of the video conferencing market. After the release of the first browsers with WebRTC support in 2013, the potential number of video conferencing terminals around the world immediately increased by 1 billion devices. In fact, each browser has become a video conferencing terminal that is not inferior to its hardware counterparts in terms of communication quality.

Use in specialized solutions

The use of various JavaScript libraries and cloud service APIs with WebRTC support makes it easy to add video support to any web projects. Previously, to transfer data in real time, developers had to study the principles of the protocols and use the developments of other companies, which most often required additional licensing, which increased costs. Already now, WebRTC is actively used in services like “Call from the site”, “Online support chat”, etc.

Ex-Skype users for Linux

In 2014, Microsoft announced that it was ending support for the Skype for Linux project, which caused great irritation among IT professionals. WebRTC technology is not tied to the operating system, but is implemented at the browser level, i.e. Linux users will be able to see a full replacement for Skype in WebRTC-based products and services.

Competition with Flash

WebRTC and HTML5 were a fatal blow to Flash technology, which was already going through its bad years. Since 2017, the leading browsers have officially stopped supporting Flash and the technology has completely disappeared from the market. But we must give Flash its due, because it was he who created the web conferencing market and offered the technical capabilities for live communication in browsers.

WebRTC video presentations

Dmitry Odintsov, TrueConf, Video + Conference October 2017

Codecs in WebRTC

Audio codecs

To compress audio traffic in WebRTC, Opus and G.711 codecs are used.

G.711 Is the oldest high-bitrate voice codec (64 kbps), which is most often used in traditional telephony systems. The main advantage is the minimum computational load due to the use of lightweight compression algorithms. The codec has a low level of compression of voice signals and does not introduce additional audio delay during communication between users.

G.711 is supported by a wide variety of devices. Systems that use this codec are easier to use than those based on other audio codecs (G.723, G.726, G.728, etc.). In terms of quality, G.711 received a score of 4.2 in the MOS testing (a score within the 4-5 range is the highest and means good quality, similar to the quality of voice traffic in ISDN and even higher).

Opus Is a low latency encoding (2.5ms to 60ms) codec with variable bitrate support and high compression, ideal for streaming audio over variable bandwidth networks. Opus is a hybrid solution that combines the best features of SILK (voice compression, human speech elimination) and CELT (audio coding) codecs. The codec is freely available, the developers who use it do not need to pay royalties to the copyright holders. Compared to other audio codecs, Opus undoubtedly outperforms in many ways. It has eclipsed quite popular low bitrate codecs such as MP3, Vorbis, AAC LC. Opus restores the “picture” of sound closer to the original than AMR-WB and Speex. This codec is the future, which is why the creators of the WebRTC technology have included it in the mandatory range of supported audio standards.

Video codecs

The questions of choosing a video codec for WebRTC took several years from the developers, in the end they decided to use H.264 and VP8. Almost all modern browsers support both codecs. Video conferencing servers to work with WebRTC only need to support one.

VP8- a free video codec with an open license, featuring high decoding speed of the video stream and increased resistance to frame loss. The codec is universal, it is easy to implement it in hardware platforms, therefore, very often developers of video conferencing systems use it in their products.

Paid video codec H.264 became known much earlier than his brother. It is a codec with a high compression ratio of a video stream while maintaining high video quality. The high prevalence of this codec among hardware video conferencing systems suggests its use in the WebRTC standard.

Google and Mozilla are actively promoting the VP8 codec, and Microsoft, Apple and Cisco are promoting H.264 (to ensure compatibility with traditional video conferencing systems). And here a very big problem arose for developers of cloud-based WebRTC solutions, because if all participants in a conference use one browser, then the conference should be mixed once with one codec, and if the browsers are different and there is Safari / Edge among them, then the conference will have to be coded twice different codecs, which will double the system requirements for the media server and, as a result, the cost of subscriptions to WebRTC services.

WebRTC API

WebRTC technology is based on three main APIs:

• (responsible for accepting audio and video signals from cameras or the user's desktop by the web browser).
• RTCPeerConnection(responsible for the connection between browsers for “exchange” of media data received from the camera, microphone and desktop. Also, the “duties” of this API include signal processing (cleaning it from extraneous noise, adjusting the microphone volume) and control over the audio and video codecs used) ...
• RTCData Channel(provides two-way data transmission over an established connection).

Before accessing the user's microphone and camera, the browser asks for permission to do so. In Google Chrome, you can pre-configure access in the "Settings" section, in Opera and Firefox devices are selected directly at the time of access, from the drop-down list. A request for permission will appear always when using the HTTP protocol and once if using HTTPS:

RTCPeerConnection... Each browser participating in the WebRTC conference must have access to this object. By using RTCPeerConnection, media from one browser to another can even pass through NATs and firewalls. For the successful transmission of media streams, participants must exchange the following data using a transport, for example, web sockets:

• the initiating participant sends to the second participant the Offer-SDP (data structure with the characteristics of the media stream that it will transmit);
• the second participant forms an “answer” - Answer-SDP and sends it to the initiator;
• then an exchange of ICE candidates is organized between the participants, if any are found (if the participants are behind NAT or firewalls).

After the successful completion of this exchange between the participants, the direct transfer of media streams (audio and video) is organized.

RTCData Channel... Support for the Data Channel protocol appeared in browsers relatively recently, so this API can be considered exclusively in cases of using WebRTC in Mozilla Firefox 22+ and Google Chrome 26+ browsers. With it, participants can exchange text messages in the browser.

Connection via WebRTC

Supported desktop browsers

• Google Chrome (17+) and all browsers based on the Chromium engine;
• Mozilla FireFox (18+);
• Opera (12+);
• Safari (11+);

Supported mobile browsers for Android

• Google Chrome (28+);
• Mozilla Firefox (24+);
• Opera Mobile (12+);
• Safari (11+).

WebRTC, Microsoft and Internet Explorer

For a very long time, Microsoft has been silent about WebRTC support in Internet Explorer and its new browser, Edge. The guys from Redmond are not very fond of putting technologies that they do not control into the hands of users, this is the kind of policy. But gradually the matter got off the ground, tk. it was no longer possible to ignore WebRTC, and the ORTC project, derived from the WebRTC standard, was announced.

According to the developers, ORTC is an extension of the WebRTC standard with an improved set of APIs based on JavaScript and HTML5, which translated into ordinary language means that everything will be the same, only Microsoft will control the standard and its development, not Google. The set of codecs has been expanded with support for H.264 and some audio codecs of the G.7XX series used in telephony and hardware video conferencing systems. Perhaps there will be built-in support for RDP (for content transfer) and messaging. By the way, Internet Explorer users are out of luck, ORTC support will only be in Edge. And, of course, such a set of protocols and codecs dock with Skype for Business with little blood, which opens up even more business applications for WebRTC.

WebRTC, Apple and Safari

According to analysts, WebRTC for Safari is already on the way, we are waiting for it in 2017. As of July 2017, WebRTC support is announced in the mobile version of iOS 11, which will be released this fall, but for now we recommend using the Chrome browser on both iOS and macOS. Apple's implementation is likely to be expanded with support for H.264 and H.265 video codecs, as well as the AAC-ELD audio codec, which is responsible for encoding audio in FaceTime (Apple's video calling application).