What is Kamailio development?

Kamailio development involves configuring, scripting, and extending the Kamailio SIP server for carrier-grade signalling, SBC functions, and large-scale SIP routing. CelloIP deploys Kamailio as an enterprise SBC, SIP proxy, registrar, and WebRTC gateway handling millions of SIP transactions daily.

What is Kamailio used for in production?

Kamailio is used as a Session Border Controller (SBC), SIP load balancer, carrier-grade proxy, topology hiding gateway, STIR/SHAKEN attestation point, and WebRTC-to-SIP bridge with rtpengine. It is the industry standard for large-scale SIP infrastructure.

How much does Kamailio SBC development cost?

Kamailio SBC configuration and customisation typically costs $8,000–$40,000 depending on feature set. Dedicated Kamailio developers are available from $25/hr at CelloIP.

Carrier-Grade SIP Infrastructure

Kamailio SIP Server Development

Expert Kamailio development for SIP registrar, LCR routing, load balancing, fraud detection, and carrier-grade VoIP infrastructure that scales to millions of concurrent sessions.

10,000+ CPS per node. Sub-1ms routing decisions. Five-nine uptime. Active-active clusters for carriers, ISPs, and VoIP platform providers.

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10K+
Calls/sec per Node
1M+
Concurrent Call States
99.999%
Five-Nine Uptime
<1ms
Routing Decision Latency

Kamailio vs OpenSIPS

Both are excellent — here is why we recommend Kamailio for carrier-grade deployments

AspectKamailioOpenSIPS
Performance✓ C-native, 10K+ CPS single nodeSimilar baseline, slightly lower ceiling
Module Ecosystem✓ 200+ modules, active developmentGood modules, smaller contributor base
Community & Support✓ Large community, commercial backingGood community, smaller enterprise adoption
Scripting Language✓ Kamailio Script + Lua/PythonOpenSIPS Script, slightly different syntax
DB Flexibility✓ MySQL, PostgreSQL, MongoDB, RedisMySQL, PostgreSQL, Redis
LCR Module✓ Advanced, real-time carrier switchingGood, slightly fewer options
Fraud Detection✓ pike, htable, dedicated modulesSimilar modules available
High Availability✓ Native active-active clusteringCluster module, more config effort
WebRTC Gateway✓ Native WebSocket/SRTP supportAvailable, less turnkey
Recommended For✓ Carrier-grade, high CPS, complex logicLighter deployments, specific use cases

Kamailio Development Services

Every layer of the SIP routing stack — from registrar to carrier interconnect

SIP Registrar & Location Service

User registration, contact tracking, multi-domain support, and presence management. Integrates with MySQL/PostgreSQL/Redis for real-time location lookups.

RegistrarMulti-domainPresence

LCR — Least Cost Routing

Dynamic carrier selection per destination. Real-time rate evaluation, automatic failover, quality-based routing, and sub-ms routing decisions at 1000+ carrier scale.

LCRDynamic Failover1000+ Carriers

SIP Load Balancer

Stateful and stateless load distribution across Asterisk/FreeSWITCH backend nodes. Consistent hashing, least-session, and session affinity strategies.

Load BalancerConsistent HashingSession Affinity

Carrier Interconnect & Peering

Multi-carrier SIP trunk configuration, E.164 normalisation, peering setup, and transparent PSTN gateway interconnection with full failover chains.

PeeringE.164PSTN Gateway

Fraud Detection & Prevention

Real-time anomaly detection via pike and htable modules. Threshold monitoring, international call spike blocking, geo-impossible detection, and CDR audit.

pike ModuleToll FraudReal-time Block

TLS / SRTP Security

Encrypted SIP signaling via TLS 1.3, mutual certificate auth, SRTP media encryption, certificate pinning, and HSM integration for regulated environments.

TLS 1.3SRTPmTLS

Kamailio + Asterisk Architecture

Separation of signaling (Kamailio) from media (Asterisk/FreeSWITCH). Kamailio handles routing at scale; Asterisk handles IVR, recording, and conferencing.

Hybrid ArchAsteriskFreeSWITCH

Monitoring & CDR Analytics

Prometheus metrics, Grafana dashboards, SIP trace capture, ASR/PDD/ACD KPIs, real-time alerting, and CDR-based revenue analytics for operators.

PrometheusGrafanaCDR Analytics

Full Capability Set

From basic SIP registrar to carrier-grade fraud prevention and analytics

SIP registrar with multi-domain support
Location service (usrloc module)
Digest & TLS certificate authentication
LCR with 1000+ carrier routing tables
Real-time dynamic carrier failover
Quality-based and time-of-day routing
Stateful & stateless load balancer
Consistent hashing session affinity
Active-active HA cluster (10+ nodes)
WebSocket / WebRTC SIP gateway
TLS 1.3 signaling encryption
SRTP media path encryption
mTLS mutual certificate auth
Pike module — brute-force protection
htable — real-time fraud thresholds
Geo-impossible call detection
Toll fraud blocking + CDR audit
Kamailio Script + Lua/Python scripting
MySQL, PostgreSQL, Redis backends
XCAP / rich presence management
SMS over SIP (MESSAGE routing)
Prometheus + Grafana integration
SIP capture (HEP/Homer)
CDR generation & revenue analytics

Deployment Tiers

From SMB VoIP provider to Tier-1 telecom — architecture that fits your scale

SMB VoIP Provider

Up to 1K CPS

  • Single Kamailio node (HA pair)
  • SIP registrar + LCR
  • Up to 10 carrier connections
  • Basic fraud detection (pike)
  • MySQL backend
  • Standard monitoring dashboards
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Carrier / Operator

1K–50K CPS

  • Multi-node active-active cluster
  • Advanced LCR (1000+ carriers)
  • Real-time fraud detection suite
  • TLS/SRTP full encryption
  • Kamailio + Asterisk architecture
  • Redis caching for <1ms lookups
  • SIP capture + Homer analytics
  • Prometheus/Grafana dashboards
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Tier-1 Telecom

50K+ CPS / Global

  • Everything in Carrier tier
  • Geo-distributed cluster (multi-DC)
  • BGP anycast for global SIP routing
  • HSM-backed certificate management
  • Custom fraud ML integration
  • Dedicated SRE engineer on retainer
  • Sub-1ms routing optimisation
  • SLA: 99.999% five-nine uptime
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Implementation Process

From requirements to carrier-grade production in 6–12 weeks

01

Requirements & Traffic Analysis

Define call volume targets, carrier list, routing logic complexity, HA requirements, and security posture. Size hardware and cluster topology.

02

Architecture Design

Design Kamailio cluster topology, database schema (usrloc, LCR tables, dispatcher), Redis caching layer, and backend media server integration points.

03

Core Kamailio Script Build

Write and test the Kamailio routing script: registration flow, LCR logic, fraud checks, load balancer dispatch, and TLS/SRTP policy enforcement.

04

Security & Fraud Hardening

Configure pike rate limiting, htable fraud thresholds, TLS certificate chain, SRTP policies, IP allowlisting, and automated block-on-alert rules.

05

Load Testing with SIPp

Simulate target CPS × 2 using SIPp scenarios. Profile bottlenecks, tune database connection pools, OS kernel params, and Kamailio memory settings.

06

Parallel Migration & Go-Live

Run new cluster alongside existing infra. Gradually shift 10% → 50% → 100% traffic. Monitor ASR/PDD/CPS in real-time. Full rollback available throughout.

Why CelloIP for Kamailio

Carrier-scale production experience — not tutorial-level expertise

Carrier-Scale Track Record

12+ years deploying Kamailio handling billions of calls. We have debugged complex SIP edge cases most engineers never encounter.

Fraud Prevention Expertise

Our toll fraud systems have saved clients hundreds of thousands in avoided fraudulent traffic. Battle-tested in live carrier networks.

Kamailio Community Contributors

Our engineers are active in the Kamailio open-source community. We know the roadmap, the quirks, and the undocumented behaviours.

Full Stack Integration

Kamailio does not operate alone. We integrate with Asterisk, FreeSWITCH, billing systems, and CDR platforms as a complete stack.

Script-Level Customisation

Deep Kamailio Script, Lua, and Python module expertise means routing logic that matches your exact business rules — no compromises.

24/7 Post-Launch SRE Support

On-call engineering for production incidents. Quarterly optimisation reviews. Scaling support as your CPS grows.

10K+
CPS per node
99.999%
Five-nine SLA
<1ms
Routing latency
200+
Kamailio modules mastered
12+
Years production VoIP
0
Vendor license fees

Frequently Asked Questions

What is Kamailio and how does it fit into a VoIP architecture?
Kamailio is a stateful SIP proxy and registrar that sits at the routing core of carrier and enterprise VoIP networks. It handles SIP REGISTER requests to track subscriber locations, routes incoming calls based on complex business logic and LCR, and acts as the intelligent SIP traffic director. Unlike a media server (Asterisk handles IVR, recording, conferencing), <a href="https://www.kamailio.org/" target="_blank" rel="noopener noreferrer" style={{color:"#f97316"}}>Kamailio</a> focuses purely on signaling and routing. The typical architecture: endpoints register with Kamailio → Kamailio routes calls to Asterisk/FreeSWITCH backends → media flows directly between endpoints.
Kamailio vs OpenSIPS — which should I choose?
Both are excellent open-source SIP servers. Kamailio is generally faster, has a larger community, more modules (200+), and stronger commercial backing. <a href="https://opensips.org/" target="_blank" rel="noopener noreferrer" style={{color:"#f97316"}}>OpenSIPS</a> is lighter and sometimes preferred for specific smaller deployments. For carrier-grade, high-CPS, or complex routing logic deployments, we recommend Kamailio. For lightweight SIP proxy needs where you want minimal configuration, OpenSIPS can work well. We have expertise in both and can advise based on your specific traffic and feature requirements.
How many concurrent calls can a single Kamailio server handle?
On modern hardware (16-core, 64 GB RAM), a well-tuned Kamailio server handles 10,000+ calls per second (CPS) and over 1 million concurrent call states. In stateless proxy mode (no dialog tracking), a single server can push significantly higher. By deploying a stateless active-active cluster, you scale linearly — 10 nodes handle 100,000+ CPS. The limiting factor is usually database lookup latency, not Kamailio itself, which is why Redis caching is critical for sub-millisecond routing at carrier scale.
How does Kamailio integrate with Asterisk or FreeSWITCH?
Kamailio acts as the SIP front-end and routing engine; Asterisk/FreeSWITCH act as the media backend. SIP endpoints register with Kamailio. Kamailio applies LCR, fraud checks, and routing decisions, then dispatches calls to the appropriate Asterisk instance using the dispatcher module. Asterisk handles IVR, call recording, conferencing, and feature logic. This separation allows Kamailio to scale routing independently of media processing — you can have 5 Kamailio nodes routing to 50 Asterisk servers.
How does Kamailio handle toll fraud detection?
Kamailio's pike module tracks REGISTER and INVITE rates per source IP and blocks IPs that exceed configured thresholds (e.g. >10 failed auth attempts in 60 seconds). The htable module enables real-time counters per subscriber — if a user suddenly makes 50 concurrent international calls when their normal is 2, calls are blocked and an alert fires. We layer additional detection: geo-impossible call origin (subscriber registered in UK, calling from Nigerian IP), time-of-day anomalies, and destination prefix monitoring (sudden spike to premium-rate numbers).
What monitoring is included in a Kamailio deployment?
We instrument Kamailio with the statistics module and export metrics to Prometheus. Grafana dashboards display: CPS (calls per second), active calls, ASR (answer seizure ratio), PDD (post-dial delay), ACD (average call duration), SIP response code distribution (4xx errors, 5xx failures), and per-carrier quality metrics. SIP capture via HEP protocol feeds into Homer for full SIP trace replay — invaluable for debugging complex call failure scenarios. Alerts trigger on critical thresholds (ASR drop, CPS spike, database latency) with PagerDuty or email notification.

Related VoIP Services

Complete the carrier-grade VoIP stack

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Further Reading

Related Blog Posts

Configuring Kamailio as an Enterprise Session Border Controller

17 min read

OpenSIPS 3.x Load Balancing: SIP Traffic Across Asterisk Clusters

14 min read

WebRTC to SIP Gateway Architecture

15 min read

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Whether you are a VoIP provider scaling to millions of minutes, a telecom needing carrier interconnect, or an operator replacing a legacy PBX with a modern SIP core — CelloIP delivers production Kamailio infrastructure that performs.

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