WebRTC & real-time comms R&D
Open source / R&DAuthor · 2024 – 2025
A side track exploring WebRTC media servers and cross-platform SIP — open-sourced as a Janus gateway wrapper and a reusable Linphone module for React Native.
Alongside the production telephony work, I kept a parallel research track open: what would calling look like built on WebRTC media servers instead of classic Asterisk, and could the SIP stack be made reusable across platforms? Two open-source repos came out of it.
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janus-react-wrapper — a React + TypeScript wrapper over the Janus WebRTC gateway, for browser-native calls routed through a Janus media server.
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react-native-linphone — an Expo native module wrapping the Linphone SIP SDK, so a React Native app can do real SIP calling without a separate native rewrite per platform. This one fed directly back into the mobile call work.
R&D in the honest sense: not everything shipped, but it mapped the territory — where WebRTC wins, where SIP is still the right tool, and how much of a SIP client you can reasonably push into a cross-platform layer.