Enterprise-Grade SIP Server Development

OpenSIPS is an open-source SIP server for routing VoIP traffic. It’s used for SIP proxy, load balancing, NAT traversal, presence, and more.

OpenSIPS is optimized for SIP signaling and control at scale, while FreeSWITCH/Asterisk focuses on media processing like conferencing and IVRs.

Yes, it can manage thousands of simultaneous calls and is ideal for carrier-grade applications.

Absolutely — it supports TLS, IP blocking, DoS detection, and SIP topology hiding.

Yes, we design OpenSIPS-based SIP load balancers with failover, redundancy, and traffic distribution.

Yes, we offer full lifecycle support, security patches, monitoring, and module updates.

Definitely ,  we integrate it with CRMs, custom dashboards, and third-party APIs via REST or event-based systems.

For media-heavy features, we integrate OpenSIPS with FreeSWITCH or Asterisk for seamless audio handling.

Yes ,  our platforms support multiple clients with billing, access control, and isolated call routing.

Yes, we offer tools for real-time SIP trace, call state, logging, and usage analytics.