DTLS-SRTP ENCRYPTED

Enterprise-Grade
WebRTC Infrastructure

Secure, scalable, and integrated voice communications for the modern enterprise. Replace legacy SIP trunks with browser-native technology.

View Architecture

constconnection=newRTCPeerConnection(config);

// Mandatory Encryption

connection.addTransceiver('audio',{ dtls: true });

// Global Low Latency

await connection.setLocalDescription(offer);

> Connection established: 24ms latency

Uncompromising Security

WebRTC mandates encryption. We utilize DTLS (Datagram Transport Layer Security) and SRTP (Secure Real-time Transport Protocol) to ensure no voice data can be intercepted or tampered with in transit.

End-to-End Encryption
SOC2 Compliant Infrastructure
No PII Storage

Infinite Scalability

Forget hardware limitations. Our cloud-native architecture scales elastically. Whether you have 10 concurrent calls or 10,000, our global mesh of media servers handles the load automatically.

99.99% Uptime SLA
Global Edge Network
Auto-Scaling Clusters

Seamless Integration

StartACall isn't just a tool; it's a platform. Integrate voice capabilities directly into your CRM, Helpdesk, or custom internal tools using our comprehensive APIs and Webhooks.

REST API Access
Real-time Webhooks
Custom CRM Widgets

Global Low-Latency Network

We don't just route calls; we optimize the path. Our intelligent routing engine selects the nearest media server to the user, minimizing latency and jitter.

12 Regions

Data Centers

< 50ms

Average Latency

System Status: Operational

Modernize Your Voice Infrastructure

Move away from hardware PBX and legacy SIP. Embrace the future of browser-based communication.