Giacomo Vacca is a Senior WebRTC Engineer with 13 years of experience building real-time communications and telecom software. As a current engineer at SignalWire and founder of RTCSoft, he combines product-minded engineering with hands-on protocol work across SIP, RTP/RTCP, and codec stacks. His open-source contributions to flagship projects like Kamailio and FreeSWITCH show deep backend expertise—improving Redis-backed modules, enhancing SIP tracing and TLS handling, and advancing codec/RTCP behavior for mod_opus. Comfortable operating between low-level telecom internals and scalable WebRTC services, he brings a practical knack for making systems more resilient when external dependencies are flaky.
Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
Role in this project:
Back-end Developer
Contributions:51 commits, 18 PRs, 13 pushes in 6 years 3 months
Contributions summary:Giacomo contributed to the core functionality of the Kamailio SIP server. Their work focused on improving the `ndb_redis` module by adding the ability to initialize the module even if Redis is unavailable, enhancing flexibility. They also modified the `sipcapture` module, accepting a list of headers for callid and adding support for HEPv3 in `siptrace`. Additionally, the user refactored query string parameter setting in the `http_async_client` module and made several improvements to the TLS-related parameters.
FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a Raspberry PI to a multi-core server, FreeSWITCH can unlock the telecommunications potential of any device.
Role in this project:
Back-end Developer
Contributions:7 commits in 2 years 1 month
Contributions summary:Giacomo contributed to the FreeSWITCH project by implementing and modifying core functionalities. The contributions focused on areas such as payload code handling, SIP header processing within the mod_sofia module, and RTCP handling. Additionally, the user made changes related to codec implementation, particularly for the mod_opus codec, including bitrate negotiation and statistics logging. Furthermore, the user added a new call cause to the core channel library.
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