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I will develop and fix webrtc video audio call and socket io apps
Rasel Hossain
À propos de ce service
Are you facing connection drops in Socket.io, or issues with WebRTC video/audio calling? Or do you need a real-time communication feature built from scratch? You're in the right place!
I specialize in building, fixing, and optimizing real-time applications using WebRTC and Socket.io. From small bugs to full-scale video calling systems, I can handle it efficiently and professionally.
Services I Offer:
- WebRTC setup & fixes (video/audio calls, screen sharing, permissions)
- Socket.io real-time chat & connection issue fixing
- STUN/TURN server setup (Coturn, Twilio, Xirsys)
- Real-time features (chat, notifications, typing status, online users)
- Deployment & server setup (Nginx, SSL, VPS, AWS)
️ Tech Stack:
JavaScript, TypeScript, Node.js, express.js, nestjs, Socket.io, WebRTC, Laravel, PHP, MySQL, Linux, Nginx
Why Me?
- 5+ years of experience in full-stack development
- Clean, scalable, production-ready code
- Fast delivery for critical fixes
- Good communication & support after delivery
Please drop me a message with your project details before placing an order so we can discuss the requirements and technical architecture!
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Découvrez Rasel Hossain
Rasel Hossain
Full Stack Web and Mobile App Developer
- DeBangladesh
- Membre depuismars 2019
- Dernière commande4 années
Langues
Anglais
Hello, I'm Rasel Hossain, a Full Stack Developer with 5+ years of experience.
My expertise includes PHP, Laravel, Node.js, Express.js, React, React Native, Next.js, NestJS, MySQL, AWS, VPS, WebRTC, and Socket.IO.
I build web applications, REST APIs, mobile apps, real-time chat systems, video calling platforms, and cloud solutions.
Services:
✔ Laravel & Node.js Development
✔ Bug Fixing & Troubleshooting
✔ REST API Development
✔ React & Next.js
✔ React Native Apps
✔ NestJS Backend
✔ WebRTC & Socket.IO
✔ AWS & VPS Deployment
✔ MySQL Optimization
Let's build something great together.
Mon portfolio
FAQ
Do I need an SSL certificate (HTTPS) for WebRTC audio/video calls to work?
Yes, absolutely. WebRTC requires a secure environment (HTTPS/WSS) to access camera and microphone permissions in modern browsers. If you don't have SSL set up, I can configure it for you on your server as part of the hosting setup.
Why are my users unable to connect to video calls on mobile networks or behind strict firewalls?
This usually happens when a peer-to-peer (P2P) connection fails because of strict symmetric NATs or firewalls. To solve this, a TURN server is required. I can install and configure a dedicated Coturn server or integrate third-party services like Twilio/Xirsys to fix this issue.
Can you fix real-time application issues on an already live production server?
Yes, I can debug and fix issues on production servers. However, to ensure zero downtime for your active users, I prefer replicating the issue on a staging/local environment first, fixing it there, and then safely deploying the tested code to your live server.
What do you need from me to start working on the project?
I will need access to your source code repository (GitHub/GitLab), server access (SSH/Panel) if deployment or configuration is needed, and a clear description or video screen recording of the bug or feature you want to implement.
