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Open-Source Platforms

Kamailio Development

Kamailio handles more concurrent SIP sessions per CPU than any other open-source SIP server — if you know how to write its configuration. We build Kamailio deployments for load balancing, SIP registration, least-cost routing, carrier SBC duty, and WebSocket gateways for WebRTC. Our configuration style favors modular kamailio.cfg files, htable caching for hot paths, dispatcher for failover, and KEMI (Lua, Python, or JavaScript) when the routing logic gets complex enough that raw KSR becomes hard to review. We also write custom Kamailio modules when upstream doesn't cover a specific carrier requirement.

Scope This Work → See All Services

Who it's for

  • ITSPs routing millions of SIP calls per day across multiple POPs
  • Platforms needing SIP registration for 100k+ endpoints
  • WebRTC products needing SIP-over-WebSocket gateway
  • Carriers deploying Kamailio as an SBC for topology hiding and transcoding offload

Our approach

  1. 1Start with a clean kamailio.cfg — no copy-paste from the wiki
  2. 2Use dispatcher with probing_mode=1 for carrier-aware failover
  3. 3Cache hot lookups in htable to avoid hitting the database per INVITE
  4. 4Use TLS and SRTP by default for new deployments — never trust the network
  5. 5Version-control the config and deploy via CI so every change is reviewable

What you get

Modular kamailio.cfg split into request_route, branch_route, and failure_route files

Dispatcher configuration with tuned probing intervals and failover priorities

MySQL, PostgreSQL, or Redis backend for routing, acc, and usrloc

WebSocket gateway configuration for WebRTC when needed

KEMI scripts (Lua or Python) for logic that outgrew raw KSR

Grafana dashboards wired to Kamailio's CORE and Prometheus exporter

Common questions

Ready to build on carrier-grade voice?

Talk to a VoIP engineer — not a salesperson.