José Millán is a WebRTC senior engineer with 13+ years building real-time communications systems and a track record of shipping foundational open-source projects like mediasoup and JsSIP. He co-authored RFC7118, contributed to the W3C ORTC spec, and led core implementation work on a high-performance SFU used in production video conferencing. José blends low-level C/C++ and network programming with JavaScript/TypeScript front-end integrations, evident from contributions across mediasoup-client, libmediasoupclient and SIP stacks such as baresip and SIP.js. His career spans startups and product teams (Around, Miro, Restream) focused on scalable multiparty VoIP and WebRTC experiences for enterprises. Known for practical protocol expertise, he often works on subtle interoperability and monitoring improvements (e.g., stats and transport handling) rather than only feature UI work. Based in Spain, he pairs academic telecoms training with hands-on engineering across SIP, WebRTC, GNU/Linux and real-time media.
13 years of coding experience
16 years of employment as a software developer
Degree in Telecommunications, specializing in Telematics, Degree in Telecommunications, specializing in Telematics at Universidad del País Vasco/Euskal Herriko Unibertsitatea
Contributions:34 reviews, 215 commits, 36 PRs in 3 years 5 months
Contributions summary:José primarily worked on refactoring the `libmediasoupclient` library, modifying and restructuring the code for improved functionality and maintainability. They implemented and updated code related to transport, producer, and consumer modules, making changes to handle functionalities related to SDP and RTP parameters. The changes indicate a focus on enhancing the core components of the mediasoup client-side library.
Contributions:32 reviews, 582 commits, 107 PRs in 10 years 2 months
Contributions summary:José contributed to the `jssip` repository, a JavaScript SIP library. Their work focused on improving the 'tryit' demo, notably removing the requirement for full SIP URIs and implementing options for WebSocket URI settings. They also updated the code to transition from `Peerconnection00` to `RTCPeerconnection` and made various code changes related to session events and DTMF (Dual-Tone Multi-Frequency) support, demonstrating expertise in WebRTC and SIP signaling. These changes included refactoring session management and fixing issues related to request handling and credential reuse.
freeswitchjavascriptsip-libraryjssipsip
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