WebRTC is a real-time communication technology that lets you make audio and video calls directly from your browser. It is well-known for providing superior security and propelling WebRTC communication to the top of the pack. However, by building your communication app on top of WebRTC, you can leverage the underlying technology’s inherent security mechanisms, ensuring that your chat and video calling API provides the best protection to its users.
In this blog, we will discuss WebRTC security and see how this technology achieves the level of protection and privacy that any modern communication app requires.
Let’s start with WebRTC’s Security Mechanisms:
The WebRTC’s Security Mechanisms
WebRTC puts a high priority on security, using the appropriate security mechanisms at its core. The following components, however, include security mechanisms:
- SRTP
- Session (TLS)
- Session (DTLS)
WebRTC Signaling
WebRTC uses inherently encrypted protocols such as TLS or QUIC for signaling rather than non-encrypted signaling channels. Non-encrypted channels can be used, but they can be restricted by certain browsers or request access to the user’s microphone and camera at the start of each session. Therefore, we do not recommend WebRTC-based deployments over unencrypted signaling channels.
WebRTC Encryption is a vital step. At the same time, when it comes to authentication and identity management, you can use whatever mechanism that works best for you. It applies to any application you create, mobile or desktop; the only requirement is encrypted signaling.
WebRTC Media Security
WebRTC’s primary goal is media communication. Data sent over WebRTC peer-to-peer is always encrypted. All audio and video streams are sent using SRTP. Data communication channels, like audio and video channels, use DTLS and are always encrypted.
DTLS-SRTP is required to obtain a shared private key for an SRTP session. This mechanism stops fraudsters from acquiring the key. SDES is also prohibited to increase security, which means that keys cannot be sent in their initial form even over secure channels. WebRTC prefers the “zero trust” approach to signaling channels.
The WebRTC-based app security architecture states clearly that the following security mechanisms are always mandatory:
- Implementations should support SRTP.
- DTLS and DTLS-SRTP for SRTP keying.
- Implementations should support SCTP over DTLS.
- All media channels should be secured via SRTP and SRTCP.
- Media traffic should not be sent over plain (unencrypted) RTP or RTCP.
- Implementations should not negotiate cypher suites with NULL encryption modes.
- Every media channel should support DTLS-SRTP.
- WebRTC implementations should not provide or choose SDP Security Descriptions.
- An SRTP MKI should not be used.
- All data channels should be secured via DTLS
The media encryption keys are generated during the handshake and are only visible to the two connection participants. The only risk is a man-in-the-middle attack on the signaling server. When your application makes use of a media server, it becomes a trusted entity. As a result, the server terminates the encryption.
Camera and Microphone Access
When an application needs access to the microphone and camera, it generally asks the user explicitly for permission. Access to the microphone and camera is never automatic unless this is your own app.
Some browsers save the user’s access permissions between sessions, while others request access with each new session. Furthermore, the app cannot edit the access request popup. It stops fraudsters from modifying the dialogue windows that trick users into making incorrect decisions.
Screen Sharing Permission
A Webrtc video call app/webrtc video chat app will ask the user’s permission to share their screen in any browser. Every browser has a dedicated dialogue window for this purpose. These permissions are never saved until the next session, so the user must grant permission each time.
Local IP Addresses and Privacy
Despite its benefits, WebRTC API had a privacy issue that raised many concerns: it exposed the browser’s local IP address to send it over the signaling channel with JavaScript. This problem has now been resolved by the addition of mDNS.
WebRTC Security in Open-Sourcing and Standardization
One of the key aspects of the WebRTC protocol is that it is open-source and standardized. Standardization increases development time and resource consumption, which might explain why there is no standard WebRTC specification yet, despite being in the draft stage for a long time.
When it comes to Web video calling security, the standardized approach is preferable to the proprietary one. You may even prioritize security over business needs. Furthermore, WebRTC was designed to be run in a browser, which immediately implies a higher level of danger and malicious attacks.
WebRTC Security-Supported Browsers
Browser security is a primary concern. Browsers serve as a barrier between users and the vast area of the Internet, so they must safeguard users against a wide range of threats. As a result, every major browser is working hard to improve its security features.
When it comes to WebRTC security, consider the following:
- Browsers release updates regularly. Therefore, WebRTC and API Integration are always up to date and, as a result, the most secure. At the same time, application vendors must keep up with browser updates while ensuring regular product updates.
- As Webrtc video calls and Webrtc video chat rely on browsers, their security mechanisms are far more reliable than those implemented in any VoIP product.
- The results of the WebRTC fuzzing experiment, which used fuzzed messages to detect security vulnerabilities that were later fixed.
When we talk about Web browser video call API, we’re talking about something that billions of users have access to at any given time. Their security measures and mechanisms are built to withstand such loads and prevent such threats. WebRTC sits on top of these strong security measures, providing similarly secure service.
The Security of Proprietary Products Is Lower
If you want to use a proprietary solution instead of WebRTC for your video communication product, keep in mind that the level of security will be lower. There are several reasons for this, including
- Any proprietary solution is created and developed by a small team of people. It will never compare to an open-source product that has been analyzed, tested, tried, and examined by communities far larger than any QA team.
- Any company that creates its own proprietary tool or product is primarily concerned with making a profit. It has a budget that it must stick to. In such cases, security may become less important than user-facing features, which receive a larger share of investment.
- Without WebRTC, you must find a way to run your product in a browser. Some of the possible methods are not entirely safe and may result in security issues with your product.
- It’s not a good idea to create your own security solutions. In this case, the security of the entire app and all of its users’ data is reliant on your team’s professionalism, and any small error can lead to a massive disaster.
At one point, many developers considered running media solutions without restrictions, but this proved to be a bad idea. WebRTC uses the most effective approach for today’s networks: “zero trust.” Each component’s security mechanism must be implemented independently.
What You Can Do to Enhance WebRTC-Based App Security?
When you build a webRTC video chat app with exceptional security mechanisms, your app is as safe as you design it to be. You can ensure high security by doing the following:
- Implement security mechanisms, in addition to the inherent security of WebRTC, to meet the needs of your app. Remember how we talked about authentication and identity management? This is where you can add an extra level of security.
- Ensure that WebRTC encryption and security mechanisms are properly implemented within your application.
- Pay close attention to mistakes that can lead to major security breaches, such as using an outdated server or not conducting enough testing.
Wrapping Up
WebRTC is a game-changing technology that will enable us to do incredible things in the near future. It’s not just voice or video calling, but it can also be used for cloud gaming, unified communications, and other purposes. At the same time, the Internet is a vast place full of threats and risks, and the threat landscape is constantly changing. As a result, we should constantly seek new ways to protect ourselves. Data theft is one of the most serious risks associated with online communication. Anyone who uses the internet faces a big risk. The encryption and security features of WebRTC can help protect you from these threats.