Question 1

What is OpenSIPS and what is it used for?

Accepted Answer

What is OpenSIPS: it is an open source SIP proxy and server built for high-performance signaling in carrier and service-provider environments. OpenSIPS SIP proxy explained: it routes SIP messages between endpoints, handles load balancing across media servers, manages registrations, and enforces routing and security policies. OpenSIPS open source licensing under GPL means it can be freely deployed and extended, and its modular architecture allows specific capabilities to be added without affecting core performance.

Question 2

What is the difference between OpenSIPS and Kamailio?

Accepted Answer

OpenSIPS vs Kamailio: both are high-performance SIP servers descended from the OpenSER project, and both are used in carrier-scale deployments. The OpenSIPS Kamailio difference is mainly one of focus and ecosystem. OpenSIPS emphasises a built-in control panel, a scripting language optimised for readability, and a module set suited to service-provider platforms. Kamailio has a broader module library, stronger IMS/VoLTE support, and wider adoption among traditional telecom operators. Which is better OpenSIPS or Kamailio depends on your use case: OpenSIPS for service-provider and CPaaS infrastructure, Kamailio for operator-grade IMS and carrier interconnect.

Question 3

What throughput can OpenSIPS handle?

Accepted Answer

OpenSIPS performance is among the highest of any open-source SIP server. OpenSIPS calls per second on a single node can exceed 50,000 call setups, with the exact figure depending on routing complexity, active modules, and hardware. OpenSIPS throughput benchmark results vary by configuration, but its asynchronous, multi-process architecture and efficient memory management are designed to sustain high transaction rates without degrading under load. Horizontal scaling across multiple nodes extends this further.

Question 4

What is the difference between OpenSIPS and a media server like FreeSWITCH?

Accepted Answer

OpenSIPS vs FreeSWITCH is a comparison between two complementary components, not alternatives. The SIP proxy vs media server distinction is important: OpenSIPS routes SIP signaling but does not process audio or video. OpenSIPS media handling is limited to signaling-level manipulation of SDP and headers; actual media flows separately. FreeSWITCH is a media server that terminates calls, mixes audio, handles conferencing, and runs dialplan logic. In production, OpenSIPS typically sits in front of FreeSWITCH or Asterisk, handling routing, load balancing, and registration while the media server handles the calls themselves.

Question 5

Does OpenSIPS support WebSocket and WebRTC?

Accepted Answer

Yes. OpenSIPS WebSocket support is native, implementing SIP over WebSocket per RFC 7118. OpenSIPS WebRTC support works at the signaling layer: it acts as a protocol gateway, accepting SIP-over-WebSocket connections from browser-based clients and routing them to traditional SIP infrastructure. OpenSIPS SIP over WebSocket allows WebRTC clients to register, place, and receive calls through the proxy without a separate gateway component. Media bridging between WebRTC SRTP and standard RTP is handled by an integrated relay such as RTPEngine.

Question 6

How does OpenSIPS handle fraud detection and prevention?

Accepted Answer

OpenSIPS fraud detection is handled through a combination of modules that inspect call patterns in real time. OpenSIPS anti fraud measures include rate limiting via the pike module, which tracks request rates per source IP and blocks traffic exceeding defined thresholds. The OpenSIPS pike module is the primary tool for protecting against flood attacks and scanning attempts, operating at the signaling layer before requests reach backend systems. Additional protections include concurrent call limits per user or domain, geographic restrictions, blacklisting, and anomaly scoring through the fraud_detection module, which can trigger alerts or blocks based on deviation from normal call patterns.

Question 7

Can OpenSIPS be used as a Session Border Controller (SBC)?

Accepted Answer

Yes. OpenSIPS SBC functionality is achievable through a combination of its existing modules, covering the core capabilities expected of a session border controller. OpenSIPS session border controller deployments typically include topology hiding to mask internal network structure, NAT traversal with RTPEngine for media anchoring, TLS/SRTP for encrypted transport, rate limiting, codec filtering via SDP manipulation, and access control policies. OpenSIPS SBC configuration requires assembling these modules into a coherent routing script, with the result being a deployable SBC that does not require a separate proprietary appliance.

OpenSIPS Development Services

What We Build

SIP Proxy and Routing

Load Balancing

SBC Functionality

Registrar and Location Service

Fraud Detection and Prevention

Multi-Tenant Architecture

Use Cases

VoIP Service Providers

Wholesale Carriers

CPaaS Backends

Enterprise SIP Infrastructure

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Related Services

Kamailio Development

FreeSWITCH Development

WebRTC Development