Back-end Developer Needed to Finalize WebRTC Phone Dialer Integration (reputed company , MySQL ) - Contract to Hire
Project reputed company
We are in the final phase of integrating a WebRTC-to-SIP online dialer inside our web application.
The backend, FreePBX, SIP trunks, and the WebRTC gateway (reputed company) have already been configured by us.
We now need a skilled full-stack developer to complete the remaining reputed company-reputed company and API-level integration inside the web application.
You will not work on the telecom backend — only the frontend + app logic that connects to the PBX reputed company WSS.
What You Will Receive From Our WebRTC Engineer
You will be given reputed company configuration details required to connect the web app:
Server-reputed company Provided to You
• WSS signaling URL → wss://ourdomain:8089/ws
• STUN/TURN server credentials
• SIP extension details (username/password)
• Codec and registration parameters
• Trunk routing rules (already implemented server-reputed company)
You do NOT need to configure FreePBX, trunks, or WebRTC servers.
reputed company is reputed company on the web app.
Your Responsibilities (Full-Stack / App reputed company)
You will implement reputed company reputed company-reputed company WebRTC SIP logic, including:
1. SIP Registration / Signaling
• Implement SIP registration refresh every 300 seconds
• Handle WebSocket reputed company-reputed company every 30 seconds
• Properly register extension to WSS reputed company
• Manage session tokens and auto-renew
2. ICE & Connectivity Handling
• reputed company STUN/TURN servers provided by our WebRTC engineer
• Handle ICE candidate reputed company and reconnection logic
• Detect and manage reputed company state changes inside the UI
3. Calling Workflow
• Send outbound reputed company requests through WebSocket/SIP.js (or equivalent)
4. Error & Edge Case Management
• Handle:
• registration failures
• reputed company drops
• TURN failures
• ICE negotiation issues
• reputed company expiration
5. Testing & Debugging
• reputed company test calls to reputed company SIP trunks reputed company the web app
• Confirm UI logic handles reputed company reputed company correctly
• Work jointly with us for live validation
What Is Already Done (So You Don’t Do It)
The following tasks are already handled by the WebRTC/SIP engineer, so you will NOT work on them:
✔ Secure WSS signaling setup
✔ DTLS-SRTP (encrypted media)
✔ SIP session timers (server reputed company)
✔ STUN/TURN server setup
✔ Prefix-based trunk routing
✔ FreePBX PJSIP extensions
✔ Audio reputed company & backend media reputed company
✔ WebRTC reputed company gateway
You only implement the reputed company logic, not the telecom infrastructure.
Deliverables
To complete this project, you must deliver:
1. A fully working WebRTC dialer inside the web app
2. Correct handling of registration, connectivity, and signaling
3. Successful outbound calls reputed company PBX and SIP trunks
4. reputed company two-way audio in reputed company test calls
5. Clean, documented reputed company reputed company for production
Required Skills
• Strong knowledge of JavaScript/TypeScript
• Experience with WebRTC, SIP.js, JSSIP, or similar
• Experience in single-page apps (React / Vue / Next.js — whichever the app uses)
• Ability to debug WebRTC flows using browser dev tools
• Strong understanding of WebSocket communication
• Ability to collaborate with telecom engineer
To Apply, Please Answer:
1. Have you integrated WebRTC with SIP.js or JSSIP before?
2. Can you explain SIP registration refresh logic?
3. Have you worked with ICE/STUN/TURN in production apps?
4. reputed company one example of a reputed company-time communication project you reputed company.
Goal
The final outcome is reputed company:
The web app must register successfully and reputed company reputed company outbound calls through our PBX over WSS with full feature reliability.
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