Urgent WebRTC Video Conferencing Engineer
Budget: $45 – $50 USD
I’m building a browser-based video-conferencing web app and need an experienced WebRTC engineer who can take it from proof-of-concept to a production-ready solution on desktop browsers.
Core scope
• Set up peer-to-peer / SFU signalling so two or more users can join the same room from Chrome, Firefox and Edge without plugins.
• Implement reliable server-side or cloud recording so each session is stored and downloadable in a standard format (mp4/webm).
• Optimise media streams for low latency and solid audio/video sync even on moderate bandwidth.
Tech expectations
Most of the stack is still flexible, but you should be comfortable working with typical WebRTC tooling such as mediasoup, Janus, Jitsi, Kurento, simple-peer, Socket.io or equivalent signalling layers, plus modern JavaScript/TypeScript on the front end. The back-end can run on Node.js, Go or Rust—choose what lets you reach stable recordings fastest.
Acceptance criteria
• Two-party and multi-party calls run for 30+ minutes with <300 ms end-to-end latency.
• A complete recording is produced, automatically saved, and playable in any HTML5 player.
• Clear, commented code and short setup documentation so another developer can deploy in one command.
This is time-sensitive and I’d like to begin immediately, so please highlight similar WebRTC video projects you’ve shipped and let me know how quickly you can deliver a first working room with recording.
Core scope
• Set up peer-to-peer / SFU signalling so two or more users can join the same room from Chrome, Firefox and Edge without plugins.
• Implement reliable server-side or cloud recording so each session is stored and downloadable in a standard format (mp4/webm).
• Optimise media streams for low latency and solid audio/video sync even on moderate bandwidth.
Tech expectations
Most of the stack is still flexible, but you should be comfortable working with typical WebRTC tooling such as mediasoup, Janus, Jitsi, Kurento, simple-peer, Socket.io or equivalent signalling layers, plus modern JavaScript/TypeScript on the front end. The back-end can run on Node.js, Go or Rust—choose what lets you reach stable recordings fastest.
Acceptance criteria
• Two-party and multi-party calls run for 30+ minutes with <300 ms end-to-end latency.
• A complete recording is produced, automatically saved, and playable in any HTML5 player.
• Clear, commented code and short setup documentation so another developer can deploy in one command.
This is time-sensitive and I’d like to begin immediately, so please highlight similar WebRTC video projects you’ve shipped and let me know how quickly you can deliver a first working room with recording.