Driansh Softtech

VoIP Development

Kamailio development services for high-capacity SIP networks

We build Kamailio SIP proxies, load balancers and registrars that route, secure and scale your voice traffic, for carriers, VoIP providers and platform teams.
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Free consultation with our engineers

What we build with Kamailio

Kamailio development is how we give your VoIP platform a fast, flexible SIP core. We write routing scripts and custom modules, and configure Kamailio as a proxy, registrar, load balancer or edge server. We work with carriers, VoIP providers and SaaS product teams alike.

Kamailio handles signaling, not media. So we pair it with FreeSWITCH, Asterisk or RTPengine, and connect it to your databases and APIs. The result is a network that keeps up as users and call volumes grow.

Technology we work with

  • Kamailio
  • SIP
  • RTPengine
  • FreeSWITCH
  • Asterisk
  • WebRTC
  • Lua
  • Python
  • MySQL
  • PostgreSQL
  • Redis

What we offer

Kamailio development services we offer

Routing script development

Write clear Kamailio routing logic for number lookups, least-cost routes and per-customer rules.

Load balancing & failover

Spread calls across media servers with the dispatcher module, so a failed node never drops service.

SIP edge security

Block scanners, floods and toll fraud with rate limits, IP filtering and authentication at the edge.

WebRTC gateway

Let browsers register over WebSocket and bridge them to your SIP network through RTPengine.

Custom modules & APIs

Extend Kamailio with new modules or HTTP and JSON-RPC calls to your own services.

Tuning & support

Profile, tune and monitor existing Kamailio clusters, and keep them on supported releases.

Scaling your SIP network?

Tell us about your traffic and setup. A Kamailio engineer will review it and suggest an architecture.
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What we build

Solutions we build with Kamailio

Class 4 softswitch

Wholesale routing between carriers with failover and rate-based routes.
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Class 5 softswitch

Retail subscriber registration and calling at provider scale.
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Multi-tenant IP PBX

A SIP front end that directs each tenant to the right PBX node.
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Unified communications

Presence, messaging and calls on one SIP platform.
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Audio & video conferencing

Browser and SIP users joining the same conference rooms.
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Why Driansh

Why teams choose Driansh for Kamailio

Signaling plus media

We build the media side too, so Kamailio and your FreeSWITCH or Asterisk servers work as one system.

Tested under load

We load-test routing and failover before launch, so you see how the platform behaves at peak.

You own the code

Routing scripts, modules and documentation are handed over, with no lock-in to our team.

Built to grow

Stateless designs and shared databases let you add servers when traffic rises, without a redesign.

How we work

From idea to live platform in four steps

  1. 01

    Discover

    We review your requirements, traffic and existing stack, then agree scope, milestones and cost.

  2. 02

    Design

    We plan the architecture, integrations and failover so the platform fits your growth plans.

  3. 03

    Build & test

    We develop in short sprints with regular demos, then load-test and QA before go-live.

  4. 04

    Deploy & support

    We launch on your servers or cloud and stay on for monitoring, fixes and upgrades.

Frequently asked questions

Get in touch

Scaling your SIP network?

Tell us about your traffic and setup. A Kamailio engineer will review it and suggest an architecture.

Tell us about your project

Share a few details and the right engineer will get back to you.

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