WebRTC-to-SIP Gateway Development

I need a seasoned engineer to build and fine-tune a WebRTC-to-SIP gateway that lets browser clients place and receive calls on my existing SIP infrastructure. The core goals are seamless audio and video negotiation, robust NAT traversal, and production-grade security.

Here’s what I’m expecting:
• A working gateway service built from proven components (for example Janus, FreeSWITCH, Kamailio, or any combination you recommend) that accepts WebRTC calls over WebSocket and bridges them into standard SIP trunks.
• Full support for ICE, STUN/TURN, DTLS-SRTP, and proper codec handling so calls connect reliably across networks and devices.
• Clear, well-commented configuration files, deployment scripts (Docker or bare metal), and a concise technical guide that allows me to reproduce the setup from scratch.
• A short test plan demonstrating successful audio and video calls between a WebRTC browser client and a SIP endpoint.

I’ll provide server access and SIP credentials as soon as we agree on the architecture. Please outline your preferred stack, any licensing considerations, and an estimated timeline in your proposal along with examples of similar projects you’ve completed.