Sergio Murillo
Chief Architect Real Time Streaming at Dolby.io
Madrid, Community of Madrid, Spain
Summary
🤩
Rockstar🎓
Top SchoolSergio Murillo is a real-time streaming architect and engineer with 13+ years building production-grade WebRTC, SIP/IMS, and media server solutions, now leading real-time streaming architecture at Dolby.io from Madrid. He has a proven track record as a founder and senior technical leader (CTO/Tech Lead) across startups and service providers, shaping low-latency video and VOIP platforms for IPTV, Mobile TV and carrier RCS projects. Sergio combines deep protocol and backend expertise—evidenced by his transport-wide congestion control implementation in the widely used Janus WebRTC server—with practical integration work like updating libwebrtc for OBS Studio forks. Comfortable moving between standards, carrier-grade systems and consumer streaming stacks, he focuses on improving video quality and congestion resilience at scale. He holds a Master’s in Computation Science from Universidad Complutense de Madrid and brings a developer-first approach to architecture, often contributing hands-on code to open-source media projects.
13 years of coding experience
24 years of employment as a software developer
Master’s Degree, Computation Science, Mathematica, Master’s Degree, Computation Science, Mathematica at Universidad Complutense de Madrid
English
Stackoverflow
11reputation
2kreached
1answer
2questions
Github Skills (12)
c1710
ice10
c1110
sdp10
webrtc10
rtp10
rtc10
networking9
websocket9
c-language8
cprogramming-language8
google-cloud-sql6
Programming languages (13)
Github contributions (5)
CoSMoSoftware/OBS-studio-webrtc
Jan 2018 - Dec 2021
This is a fork of OBS-studio with generic support for webrtc. It leverages the same webrtc implementation most browsers use.
Role in this project:
Back-end Developer
Back-end DeveloperContributions:33 commits, 2 PRs, 20 pushes in 3 years 11 months
Contributions summary:Sergio contributed to the `cosmosoftware/obs-studio-webrtc` repository, which is a fork of OBS-studio with added WebRTC support. Their primary focus was updating the `libwebrtc` library and integrating it into the project, as evidenced by the code changes in header and source files. These changes involved modifying audio and video capture modules to support the updated library interface. They also added support for a SpankChain websocket client.
libwebrtcobsleveragesbrowserswebrtc
meetecho/janus-gateway
Dec 2017 - Jan 2018
Janus WebRTC Server
Role in this project:Back-end Developer
Back-end DeveloperContributions:21 commits, 1 PR, 9 comments in 19 days
Contributions summary:Sergio implemented and integrated the transport wide congestion control (transport-wide-cc) RTP extension within the Janus WebRTC Server. This involved modifying existing code in the videoroom plugin, rtp and ice modules to support negotiation, parsing, and transmission of transport-wide-cc information. The changes included adding new parameters, data structures, and functions related to the extension. The work enables better congestion control and improved video quality for WebRTC streams.
webrtc-serverjanuswebrtc
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