What are the best ways to configure WebRTC for unified communications?

2 Use a TURN server

A TURN (Traversal Using Relays around NAT) server is a server that relays the media traffic between the WebRTC peers when a direct peer-to-peer connection is not possible due to network restrictions or firewalls. A TURN server can improve the connectivity and reliability of WebRTC sessions, especially for UC scenarios that involve multiple participants, different networks, or mobile devices. A TURN server can also provide encryption and authentication for the media traffic, enhancing the security and privacy of WebRTC communication. You can use a public or a private TURN server, depending on your budget, resources, and requirements. However, you should always test the performance and availability of your TURN server before deploying it for UC.

3 Optimize the media quality

WebRTC uses adaptive bitrate (ABR) to adjust the media quality according to the network conditions and available bandwidth, however this may not always provide the optimal quality for UC, particularly for video communication. To optimize the media quality and reduce latency, you can employ techniques such as simulcast, which sends multiple video streams with different resolutions and bitrates so the receiver can choose the best one for their device and network. Additionally, scalable video coding (SVC) can be used to encode the video into layers of different quality and complexity that can be dropped or added as needed. Congestion control algorithms (CCAs) can also be used to monitor and regulate the media transmission rate based on feedback from the receiver and the network. Finally, codecs such as VP8, VP9, H.264 for video and Opus or G.711 for audio offer better compression, quality, and compatibility.

4 Implement security measures

WebRTC has built-in security features, such as encryption, authentication, and certificate verification, however, these may not be enough to protect your UC communication from malicious attacks. To increase security, you can use HTTPS and WSS (WebSockets Secure) for the signaling and data channels to prevent unauthorized access and tampering. DTLS-SRTP (Datagram Transport Layer Security-Secure Real-Time Transport Protocol) should be used for the media channels to ensure end-to-end encryption and authentication. Additionally, ICE (Interactive Connectivity Establishment) can be used to verify the identity and validity of the WebRTC peers to guard against man-in-the-middle attacks. It is also important to use WebRTC APIs and libraries that are updated and maintained, and follow the best practices and standards for WebRTC development and deployment.

5 Test and troubleshoot

Testing and troubleshooting your WebRTC setup regularly is essential for ensuring it works properly and smoothly for UC. You can use WebRTC samples and demos to check the basic functionality and compatibility of WebRTC in different browsers and devices. Additionally, WebRTC statistics and analytics can be used to monitor and measure the performance and quality of WebRTC sessions. Logs and traces can be used to identify and diagnose the source of errors or failures, while debugging and simulation tools can be used to simulate different network conditions and scenarios. All of these tools can help you debug and fix WebRTC problems, such as connectivity, bandwidth, or codec issues.