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Voice & VoIP

VoIP Routing & Billing

Full-stack carrier VoIP with real-time AAA billing, Least Cost Routing (LCR), multi-tier failover, STIR/SHAKEN caller authentication, and per-call CDR generation. Handle 30,000+ concurrent calls with sub-50ms authorization latency on commodity hardware.

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Architecture

How VoIP Billing Works

An incoming SIP INVITE triggers a real-time Access-Request to the billing core. Authentication validates the originating device by IP, balance is checked against the estimated call cost, LCR routing selects the best termination point, and a dial string is returned — all in under 50 milliseconds. At call end, CDR generation and atomic balance deduction complete the billing cycle.

Features

Complete VoIP Platform

Least Cost Routing
Prefix trie-based LCR selects the cheapest qualifying termination point for every call in microseconds. Weighted round-robin, quality-score, and percentage-based routing algorithms available per dial peer.
Real-Time Billing
Per-second billing with configurable increment, grace time, connection fee, and minimum time. Atomic balance deduction under mutex prevents race conditions even at 5,000 calls per second. Postpaid and prepaid modes.
Multi-Tier Failover
Automatic failover across termination points based on SIP response codes. Stop-hunting rules, capacity limits, and per-TP priority weights give fine-grained control over fallback behaviour for each destination prefix.
STIR/SHAKEN
Full RFC 8224/8225 implementation for caller attestation (A/B/C), PASSporT signing with EC P-256, and certificate chain validation. Integrates with the SHAKEN certificate authority for carrier-level caller ID trust.
CDR Generation
Every call generates a CDR written asynchronously to multiple high-availability storage backends. CDR fields include billsec, hangup cause, termination point, rate, price, exchange rate, and STIR/SHAKEN verification status.
Live Call Monitoring
Real-time active-call table shows every in-progress call with duration, current cost, origination device, and termination point. Force-hangup individual calls or entire route groups via dashboard or CLI.
Technical Specifications

Specifications

Concurrent Calls30,000 max (configurable)
Auth Latency<50ms p99
CPS Target5,000 calls per second
SIP StackIn-House Built (Proprietary)
ProtocolSIP RFC 3261, RADIUS RFC 2865
CodecsG.711, G.722, G.729, Opus
Compressed CDR Export50M records/minute
STIR/SHAKENRFC 8224/8225, ATIS-1000074

Launch Your VoIP Service

Configure your origination and termination gateways through the dashboard and start routing calls in hours. Our team provides turnkey deployment and integration support.

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