WebRTC is a free, open-source, and layered protocol existing in a complex ecosystem of application code, browsers, native devices, etc providing real-time communication (RTC) through application programming interfaces (APIs). The technology allows developers to add real-time communication and features to their applications, such as video conferencing, voice calls, and data that can be sent or shared between peers.

WebRTC for enterprise-level performance

For webRTC to provide enterprise-grade communication features, it must function within the systems built for enterprise-level performance. It must ensure

Reliability – Enterprises and the personnel involved might often face scenarios where the web browsers that they use might crash, or users might accidentally close their on-going webRTC sessions, or they might have to reload the webRTC-based collaboration session due to some error – not realizing that these actions will disconnect the session as a whole. To tackle such situations, this tool supports a process called rehydration where the state of the ongoing session is maintained on a server within the network – rather than on the endpoint. Thus, when a session is discontinued without a proper process, the server returns the state information of the disconnected session to the browser so that it can re-establish the session at the exact point that it got disconnected. WebRTC rehydration enables the networks to restart the session automatically. Apart from rehydration, features like opus codec (used to enable HD audio in webRTC) add notion to reliability as they deliver services over variable internet connections than what older codec could provide.

Scalability – WebRTC enables enterprises to provide ubiquitous communications internally between their employees as well as externally with those who wish to communicate with internal resources. However, to provide scalable connectivity, the enterprise needs to manage signaling and addressing across all the users. Furthermore, providing software development kits (SDKs) for webRTC applications that will work more seamlessly with the enterprise applications and platforms will empower the enterprise to be able to scale its effective use of webRTC.

Is WebRTC secure?

Despite its increasing popularity, WebRTC still faces a few common security concerns because of which many businesses usually will be in a dilemma whether to use it in VoIP applications or not. However, studies and researches have found otherwise and emphasized the fact that there is no better alternative to WebRTC for engaging in safe and secure real-time communications. The most common security concerns associated with the technology are regarding stealing personal and sensitive data as the technology requires access to the devices’ microphones and cameras to enable voice and video calls. Businesses fear that their audio and video conversations might be recorded remotely. A privacy breach is another risk concern where hackers and other people can track private IP addresses and other crucial device information to gain access for ad targeting.

So, is webRTC secure to use and implement? Yes. Despite the above-mentioned concerns, webRTC is still the safest technology to enable communication features into applications and platforms with its three layers of security.

1. Protocol-layer security – The three most important and mandatory webRTC encryption specifications — secure real-time protocol (SRTP), secure signaling, and secure encryption key exchange — are the protocols that encrypt the data sent over webRTC, secure the encryption keys, and protect the web server connection. SRTP encrypts the data (audio, video, and other data) sent through the webRTC session and eliminates the threat of decryption of the transmitted data without a decryption key. DTLS-SRTP is a secure encryption key exchange protocol that enables all devices communicating in the specific webRTC sessions to securely exchange the decryption key of that session and decrypt the data transmitted. Secure signaling is the signaling by the server to enable devices of a webRTC session to find each other on the internet and get connected. This is usually protected with HTTPS.

2. Browser security – The browser security standards established by the W3C along with internet security specifications are required to be met by the web browsers to ensure security. The browser security will help secure the supporting connections to a webRTC connection at large. Hence, accessing websites through HTTP or HTTPS secure connection, explicit permissions to access the users’ camera/microphone, protection of device information, explicitly getting permissions from users to share their IP addresses, etc are mandatory and crucial browser security measures.

3. Operating system security – Operating systems for desktops and mobiles have built-in security controls that are similar to browser security protocols to protect end-users. The operating system for mobiles has to be installed through the app store, which acts as an additional security layer as apps have to be evaluated and approved before they are available to the public.

Enterprise-grade webRTC service enablement

A peer-to-peer webRTC network will not address the issues concerning reliability, scalability, interoperability, and security at an enterprise-level functionality. Therefore, the best solution for enterprise-level webRTC service enablement is having a platform within the enterprise’s own architecture to provide additional and necessary services, functionality, security, and connectivity. Apart from this, the webRTC developers can follow certain practices to mitigate the security threats, whatsoever.

• Frequently updating the core webRTC libraries as old code usually is prone to more vulnerabilities.
• Swiftly fixing bugs and attending security notifications
• Removing the pieces of code that are not in use to minimize attacks
• Conduct and test RTC-specific research and issues
• Securing the supporting infrastructure like web or servers to avoid compromising the entire system.

WebRTC definitely offers a powerful paradigm for new-age communications-enabled WWW. However, if you are looking to establish a platform that enables many services that webRTC potentially provides by integrating gen-next technologies such as AI and machine learning, then Libero is your solution. Libero is a video conferencing API as a service platform powered with AI and machine learning technologies that empower the rich ecosystem of content management, real-time and on-demand video conferencing features, and is easily customized and scalable.

Contact us to know more about how we can customize Libero to meet your business and communication needs.