One common challenge when managing any kind of real-time communication service is ensuring the reliability and quality of calls across various platforms.

This talk delves into how to automate WebRTC tests for browsers and mobile devices, aiming to improve how developers and QA teams approach call quality assurance.

For this purpose, we will demonstrate a live demo testing the Janus SIP Gateway example application, giving you insights into:

- how to extract RTCP statistics

- inject audio and video into the client

- record on both ends of the call all streams

- visualise the collected metrics and calculate MOS

We will also give you an overview of the Sipfront architecture and technology stack, enabling us to automate tests at scale.