I will develop and fix video conference app based on webrtc, p2p, streaming chat app


Acerca de este Servicio
Build, optimize and scale a high performance WebRTC based video conferencing and real time chat application tailored for reliability, low latency and seamless user experience.
I deliver robust P2P and server assisted streaming solutions using modern stacks like Node.js, MERN, Laravel or Firebase integrating signaling, STUN/TURN, media handling and secure communication.
Whether you need to fix connection drops, improve media quality, or develop a full featured conferencing platform, this service is engineered for production grade performance.
What you will receive:
- Fully functional WebRTC video/audio calling system
- Real time chat with message sync and presence
- Scalable signaling server (Socket.io / SignalR)
- STUN/TURN server setup for NAT traversal
- Screen sharing & multi party conferencing
- Stream optimization (bitrate, latency, codecs)
- Debugging & performance fixes for existing apps
- Clean, documented and deployable codebase
Ready to launch or fix your WebRTC app with confidence? Send a message now and lets turn your idea into a smooth, scalable real time platform.
Conoce a Kevin Mayers
Kevin
- DeEstados Unidos
- Miembro desdeabr 2026
- Responde aprox. en:1 hora
Idiomas
Inglés, Español
FAQ
Can you fix issues in my existing WebRTC app?
Yes, I can debug connection drops, audio/video issues, latency problems, and improve overall performance.
Do you support multi-user video conferencing and screen sharing?
Absolutely, I can implement scalable group calls, screen sharing, and advanced meeting features.
Which technologies do you use?
I work with WebRTC, Node.js, MERN stack, Laravel, Firebase, Socket.io, SignalR, and FFmpeg.

