WebRTC has become ubiquitous as the technology powering various forms of video experiences online. While Session Initiation Protocol (SIP) predated WebRTC by 12 years, and had become the predominant protocol used to set up real-time media sessions between groups of users, WebRTC looked to add real-time media i.e. audio, video to every web browser without the need of a separate soft phone client.
While WebRTC has become the de-facto standard for real time communication on the Internet, SIP still sees use in some scenarios, such as bridging to phone networks (PSTN) and physical conferencing equipment.
In this talk, we talk about how we went about connecting WebRTC and SIP systems using GStreamer and SIP.js.
|Sanchayan works at asymptotic.io, an open source consulting firm based out of Bangalore and Toronto. He doesn't believe in staying in a lane and has worked on micro-controllers, Linux kernel drivers and also on libraries in the Haskell ecosystem. In recent times, he's worked on Bluetooth audio using PulseAudio and GStreamer, streaming modules for PipeWire, and media processing services using GStreamer. He's a self-declared metalhead, which probably explains why he likes Rust so much. He is a familiar face at the Bangalore Haskell and Rust meetups, which he also organises.