Blog
Engineering notes from the voice stack.
Technical writing for VoIP engineers — no fluff, no vendor puff pieces.
SBCrtpengine Deployment Guide: RTP Proxy at Scale
Deploy rtpengine as a high-performance RTP proxy: kernel forwarding module, Kamailio integration, SRTP transcoding, multi-instance clustering, and capacity planning.
ArchitectureVoIP Monitoring with Prometheus and Grafana
Build a production VoIP observability stack with Prometheus and Grafana: Kamailio stats, Asterisk metrics via snmp-exporter, RTP quality dashboards, and SLA alerting rules.
WebRTCLiveKit vs Janus vs Mediasoup: Choosing a WebRTC Media Server
Technical comparison of LiveKit, Janus, and Mediasoup WebRTC media servers: architecture, scalability, latency, SDK support, and which fits your use case best.
SIPSTIR/SHAKEN Compliance for SIP Trunking Providers
Implement STIR/SHAKEN caller ID authentication for SIP trunking: PASSporT token generation, attestation levels, STI-SP certificate management, and SHAKEN verification flow.
SIPBuilding resilient SIP routing with Kamailio dispatcher
How to configure Kamailio's dispatcher module for production-grade failover: dispatcher.list syntax, probing modes, failover priorities, and keepalive tuning.
DialerVicidial Predictive Dialer Tuning: Maximizing Agent Productivity
Tune Vicidial's predictive dialer for maximum agent productivity: dial ratio math, abandon rate compliance, MySQL query optimization, and Asterisk AGI integration patterns.
SBCSizing your SBC: a practical capacity planning guide
How to calculate concurrent sessions, CPS limits, transcoding overhead, and hardware sizing for a Session Border Controller deployment.
ArchitectureFreeSWITCH High Availability: Active-Active Cluster Setup
Configure FreeSWITCH active-active high availability using NAT traversal, shared SIP profile, PostgreSQL backend, and Kamailio load balancer for zero-downtime VoIP platforms.
WebRTCWhen to choose WebRTC vs SIP trunking for your voice app
A decision framework for product teams choosing between WebRTC and SIP trunking — including architecture tradeoffs, use cases, and a comparison table.
SIPVoIP Security: Hardening SIP Infrastructure Against Attacks
Practical SIP security hardening: authentication brute-force prevention, REGISTER flooding mitigation, TLS enforcement, fraud detection patterns, and fail2ban configuration.
WebRTCDeploying a Production TURN Server: coturn Configuration Guide
Step-by-step coturn configuration for production WebRTC: TLS setup, credential management, quota enforcement, Prometheus metrics, and multi-region deployment patterns.
SIPAsterisk Dialplan Best Practices for Production Systems
Production-grade Asterisk dialplan patterns: extension priority hygiene, macro vs GoSub, AGI integration, logging strategies, and avoiding common performance pitfalls.
SIPOpenSIPS vs Kamailio: Which SIP Proxy Should You Deploy?
A technical comparison of OpenSIPS and Kamailio covering routing flexibility, module ecosystems, performance benchmarks, and operational tradeoffs for production SIP infrastructure.