Bridging the web and telecom stacks
WebTrit is a protocol translation layer. It takes complex VoIP/telecom protocols and abstracts them into clean, simple developer APIs, allowing for seamless integration with existing telecom infrastructure.
The WebTrit stack consists of specialized services that handle different aspects of communication
Central orchestrator managing all business logic and client communication
- Dual interfaces: Developer-friendly APIs for apps + complex VoIP protocols for telecom systems
- Real-time signaling: WebSocket-based protocol (WTSP) for instant call state updates
- Multi-tenant architecture: Single deployment serves multiple organizations
- Session management: Handles user authentication, routing, and state coordination
Media gateway converting between modern web standards and traditional telecom protocols
- Protocol translation: WebRTC (browsers/mobile) ↔ SIP (telecom systems)
- Media handling: Audio/video streams, network traversal (NAT/firewall)
- Security: SRTP & DTLS encryption for all media traffic
- Scalability: Multiple instances can be deployed across regions
Translation microservice connecting WebTrit to existing VoIP and billing systems
- Standardized interface: WebTrit uses consistent API regardless of backend system
- Flexible backends: Works with PortaSwitch, 3CX, BroadSoft, Asterisk, FreePBX, custom systems
- Multi-system support: Different adapters per tenant in same deployment
- Zero migration: Integrates with existing user databases and configurations
Database (PostgreSQL)
- Session data and configuration storage
- Multi-tenant data isolation
- High-availability clustering support
Step-by-step flow showing how WebTrit connects calls from mobile apps to traditional phone systems
Mobile App → WebTrit Core → Adapter → VoIP System
- App requests system configuration for tenant
- Core routes request to appropriate adapter based on domain/tenant ID
- Adapter fetches settings from VoIP system (extensions, features, routing rules)
- System info returned to app for local configuration
App → Core → Adapter → VoIP System → Validate → Return User Data
- User enters credentials in mobile app
- Core validates session and routes to tenant's adapter
- Adapter authenticates against existing VoIP system user database
- User profile returned (balance, extensions, permissions, contacts)
- No user migration required - works with existing accounts
App ←→ Core (WebSocket) ←→ Janus WebRTC Server
- Real-time bi-directional WebSocket connection established
- App uses simple WTSP protocol commands
- Janus handles complex WebRTC negotiation (ICE, STUN, TURN)
- Media channels prepared for incoming/outgoing calls
Mobile App → Core → Janus → SIP Infrastructure → Destination
- App sends simple "make call" request via WTSP
- Core validates permissions and routes call
- Janus converts WebRTC media streams to SIP/RTP
- Call integrates with existing routing, billing, and telecom features
- Real-time status updates sent back to app via WebSocket
Scalable, multi-tenant deployment architecture designed for telecom-grade reliability
WebTrit Proxy
- Load balancing and SSL termination
- Virtual host configuration for multi-tenant deployments
- Can use nginx, Apache, or any HTTP server
- Handles static content and API routing
Backend Services
- WebTrit Core: Main application logic and signaling
- WebTrit Dialer: Browser-based calling interface
- WebTrit Janus: Media server instances (can be distributed)
- Gorush: Push notification coordination
Database Services
- PostgreSQL primary and replica instances
- Session data, configuration, and tenant isolation
- High-availability clustering options
External Services
- Push notifications: Apple APNs, Google FCM integration
- VoIP systems: Multiple concurrent connections via adapter pattern
- Billing platforms: Real-time usage tracking and CDR generation
Why this design works better than traditional softphone approaches
Simplified development experience
- No VoIP expertise required: Abstract away SIP, SDP, RTP complexity
- Modern web APIs: RESTful endpoints and WebSocket signaling
- Unified interface: Same client code works with any backend VoIP system
- Rich documentation: Clear examples without telecom jargon
Seamless integration with existing infrastructure
- Zero migration effort: Works with current user databases and PBX systems
- Multi-system support: Integrate different VoIP platforms simultaneously
- Preserve investments: Leverage existing telecom infrastructure and licenses
- Gradual modernization: Add web/mobile capabilities without replacing core systems
Enterprise-ready deployment and management
- Containerized architecture: Kubernetes-ready for modern DevOps workflows
- Built-in monitoring: Admin interfaces, metrics, and logging
- Flexible hosting: Self-hosted, cloud, or hybrid deployment options
- Horizontal scaling: Add capacity by scaling individual components
How WebTrit converts between simple developer APIs and complex telecom protocols
// What developers write
webtritClient.makeCall({ to: "+1234567890" })
webtritClient.onCallStateChanged(state => {
console.log("Call status:", state)
})// What WebTrit handles automatically
- SIP INVITE/ACK/BYE message sequences
- SDP offer/answer negotiation
- RTP/RTCP media stream management
- NAT traversal (STUN/TURN/ICE)
- Codec negotiation and transcoding
- CDR generation and billing integration
WebTrit handles the complex parts so you don't have to. Instead of wrestling with VoIP protocols, you get simple web APIs that connect seamlessly to enterprise telecom systems. We take care of the hard parts so you can focus on building great apps.
Result: You get the reliability and features of traditional telecom infrastructure with the development speed and flexibility of modern web platforms.