Building an IP Telephony Server That Never Drops a Call

Reliable voice starts with disciplined infrastructure engineering. Here is the reference architecture we deploy for finance, hospitality, and logistics clients who cannot afford downtime.

Core stack

• Asterisk + Kamailio for call control and SIP routing.
• Dual Session Border Controllers positioned at each edge to handle TLS/SRTP termination and carrier interoperability.
• Geo-redundant PostgreSQL (logical replication) to keep CDRs and provisioning data in sync.

Network blueprint

1. Segment RTP streams away from management networks and enforce QoS (DSCP 46) across switches.
2. Terminate SIP over TLS and mandate mutual certificates for remote offices.
3. Use Anycast VIPs in front of SBCs so failover is sub-second.

Operational guardrails

• Run synthetic call tests every 5 minutes from multiple POPs; alert if MOS drops below 4.0.
• Version-controlled dial plans with automated linting catch looping rules before they hit production.
• Store PCAPs for 48 hours to speed up root-cause on jitter or codec mismatches.

When to scale

If concurrents peak above 1,500 calls or you add video/contact center workloads, plan for media relay offload (SFU) plus GPU-based noise suppression. We can help map the transition without ripping out your existing investment.