Alo App is a production-grade VoIP communication platform that enables users to purchase international phone numbers and make/receive real-time voice calls over mobile networks.
The system bridges traditional PSTN telephony with modern WebRTC-based communication, enabling seamless cloud-to-mobile voice connectivity through Telnyx infrastructure.
Built for real-world usage, Alo App focuses on low-latency voice communication, reliable call routing, and scalable session management.
Responsible for the end-to-end implementation of the system, including:
- Mobile application development (React Native)
- Backend architecture (Node.js / Fastify)
- Telecom integration (Telnyx Voice APIs)
- Real-time signaling system design
- Call lifecycle and session management
- WebRTC integration and optimization
- Production deployment and stability improvements
- Purchase and manage international phone numbers via Telnyx API
- Dynamic number assignment per user
- Number lifecycle management
- WebRTC-based voice calling system
- Low-latency audio transmission
- Peer-to-peer connection establishment where applicable
- PSTN ↔ WebRTC bridging via Telnyx
- Intelligent routing between mobile clients and telecom network
- Secure call session orchestration
- Call initiation, ringing, active, and termination states
- Session persistence across network changes
- Call recovery handling for dropped connections
- WebSocket-based signaling server
- Offer/Answer exchange for WebRTC negotiation
- ICE candidate handling for NAT traversal
- Connection lifecycle synchronization between client and backend
┌──────────────────────┐
│ React Native App │
│ (Mobile Client) │
└──────────┬───────────┘
│
▼
┌──────────────────────────┐
│ Fastify Backend API │
│ (Node.js Server) │
└──────────┬───────────────┘
│
┌───────────────────┼────────────────────┐
▼ ▼ ▼
┌──────────────┐ ┌────────────────┐ ┌────────────────────┐
│ Telnyx Voice │ │ WebSocket │ │ PostgreSQL │
│ API (PSTN) │ │ Signaling │ │ + Redis State │
└──────────────┘ └────────────────┘ └────────────────────┘
- Integrated Telnyx Voice API for number provisioning and call routing
- Managed PSTN connectivity with real-time WebRTC sessions
- Implemented signaling server for peer negotiation
- Handled ICE candidate exchange and NAT traversal
- Optimized connection stability under mobile network conditions
- Designed deterministic call state machine
- Managed transitions: idle → ringing → active → terminated
- Ensured synchronization between client and backend state
- Call drop recovery mechanisms
- Network switching (WiFi ↔ Mobile data) handling
- Retry logic for failed signaling exchanges
- Session consistency across reconnects
Ensured consistent voice quality and connection stability across varying mobile network conditions.
Solved complexity of bridging browser/mobile WebRTC sessions with traditional telecom infrastructure via Telnyx.
Maintained consistent call state across:
- Mobile client
- Backend API
- Telephony provider (Telnyx)
Handled ICE negotiation and NAT traversal issues to maintain reliable peer connections.
| Layer | Technologies |
|---|---|
| Mobile | React Native |
| Backend | Node.js, Fastify |
| Telecom | Telnyx Voice API |
| Real-Time | WebRTC, WebSocket |
| Database | PostgreSQL |
| Cache | Redis |
- Production system (actively maintained and scaling)
- Real-world telecom integration in use
- Ongoing improvements in call stability and routing efficiency
This system involves live telecom integrations and production call routing logic. Core implementation details and proprietary routing logic are not publicly available due to security and operational constraints.