OpenSIPS Modules Explained | Everything Essential for Modern VoIP

OpenSIPS doesn’t come pre-packaged with assumptions. It gives you over a hundred modules and lets you decide what your VoIP infrastructure needs. These modules cover everything from user authentication to NAT traversal, call routing, presence, and media relaying. But if you’re not careful, you’ll either underuse it or overload it with unnecessary bloat.

This blog is a curated guide built from real deployment experience of what our OpenSIPS VoIP Developer actually use in production systems. We’ll break it down into categories, explain what each module is for, and help you decide when (or if) you need it.

Whether you’re optimizing business VoIP features or designing scalable routing, these are the best OpenSIPS modules for VoIP systems in 2025.

Key Modules in OpenSIPS for Every Setup

These are the backbone of most OpenSIPS configurations. Whether you’re building a SIP proxy, SBC, or Class 4 router, these modules almost always play a role.

sl – Stateless Replies

The sl module allows OpenSIPS to generate SIP replies without creating transaction state. It’s lightweight and ideal for responding to SIP OPTIONS pings, malformed requests, or requests that don’t require transaction tracking. While it doesn’t support retransmissions, it’s perfect for fire-and-forget replies that need minimal overhead.
When to use it: Always. Required for basic stateless SIP responsiveness and health checks.

tm – Transaction Management

The tm module introduces transaction state handling to OpenSIPS, enabling features like retransmission control, failure routing, and parallel forking. It also ensures that SIP messages like INVITEs and BYEs are properly tracked through timers and responses. Without tm, OpenSIPS can’t behave like a stateful proxy.
When to use it: Any time you’re handling dialog creation (INVITEs), failover logic, or SIP retries.

rr – Record-Route

This module manages SIP routing headers to ensure that subsequent in-dialog requests (BYE, re-INVITE, etc.) follow the same proxy path as the initial INVITE. rr is crucial when OpenSIPS needs to stay in the signaling path for the life of the call. It works hand-in-hand with the tm and dialog modules.
When to use it: In B2B call setups, PBXs, or any situation where OpenSIPS must remain in the call path.

maxfwd – Loop Detection

The maxfwd module protects against infinite SIP routing loops by decrementing the Max-Forwards header and rejecting messages that exceed safe thresholds. It’s a simple but critical guardrail in any SIP network, especially when multiple proxies are involved.
When to use it: Always. Prevents misconfigured proxies from creating routing loops.

usrloc – User Location Storage

This module stores the contact bindings of registered users, effectively mapping SIP URIs to IP addresses and ports. It works with the registrar module to allow SIP endpoints to register from any network location. Entries can be stored in memory or persisted via a database.
When to use it: Whenever your system supports REGISTER requests or multi-device user binding.

registrar – Registration Handling

registrar processes REGISTER requests and updates the user location database via usrloc. It also handles expiration times, contact merging, and various registration scenarios. This module makes OpenSIPS function as a SIP registrar for softphones, hardphones, or gateways.
When to use it: Essential for hosted PBX, UCaaS, or any setup where endpoints register directly with OpenSIPS.

dialog – Dialog State Tracking

The dialog module tracks the full state of SIP dialogs (INVITE to BYE), including call duration, endpoints involved, and session status. It’s essential for generating accurate CDRs, handling BYE detection, and enabling advanced call control logic. It also integrates with modules like acc and b2b_logic.
When to use it: When you need session tracking, mid-call logic, or accurate accounting records.

Routing & Load Balancing Modules

These modules shape how OpenSIPS distributes and manages call flows across multiple endpoints, carriers, or clusters. They’re essential in Class 4 switches, SIP trunking gateways, and systems that need intelligent failover or dynamic routing.

dispatcher – SIP Load Balancer

The dispatcher module enables OpenSIPS to distribute SIP messages to a set of destinations based on various algorithms like round-robin, hashing, or priority. It supports dynamic reloading and can detect failed destinations to avoid routing calls to dead nodes. Combined with health checks, it’s ideal for scaling horizontally.
When to use it: For SIP load balancing, redundancy, and traffic distribution across media servers or gateways.

drouting – Dynamic Routing Engine

drouting offers advanced routing logic using a database-driven routing table. You can set route weights, failover rules, and cost preferences, making it ideal for carrier-grade call routing or LCR (Least Cost Routing). It also supports prefix matching, fallback rules, and prioritization.
When to use it: For wholesale VoIP, Class 4 routing, or multi-carrier environments with dynamic route logic.

dialplan – Number and Logic Processor

dialplan allows inline manipulation of SIP headers, caller IDs, and dialed numbers using rules defined in a database. It’s powerful for digit translations, call normalizations, or routing decisions based on number formats. Updates can be made live without restarting OpenSIPS.
When to use it: When you need flexible number formatting, LCR, or centralized digit logic.

Security & Access Control Modules

SIP systems exposed to the public are often targets for fraud, scanning, and abuse. These modules harden your deployment and enforce who can connect and what they can do.

auth + auth_db – SIP Authentication Stack

These modules work together to validate SIP credentials using digest authentication. auth_db queries a backend database like MySQL or Postgres to check user credentials. Proper integration ensures secure login and registration for endpoints.
When to use it: For authenticating SIP users or devices. Commonly used in PBX, UCaaS, or residential VoIP systems.

permissions – ACL & IP Filtering

permissions provides Access Control Lists for filtering SIP traffic based on IP, network ranges, or domains. It’s frequently used to allow/deny calls based on carrier IPs or tenant isolation.
When to use it: To block unwanted traffic, define IP-based access, or isolate multi-tenant routes.

topology_hiding – Privacy & Header Scrubbing

This module hides internal SIP topology by rewriting sensitive headers and stripping route information. It prevents endpoint details (like internal IPs or call logic) from leaking to the outside. It’s often required in B2BUA and SIP trunking scenarios.
When to use it: To enforce SIP privacy and protect network structure, especially at network borders.

tls – Transport Layer Security

The tls module adds support for SIP over TLS, encrypting SIP signaling between endpoints and proxies. It enables secure calling over public networks and is often required for regulatory compliance or encrypted softphones.
When to use it: For secure SIP signaling, especially with mobile apps, WebRTC, or inter-carrier communication.

proto_wss – Secure WebSockets

proto_wss allows OpenSIPS to handle SIP over WebSocket Secure (WSS), the standard transport for WebRTC and browser-based SIP clients. It integrates with tls to ensure end-to-end encrypted signaling.
When to use it: When supporting WebRTC clients (e.g., SIP.js, JsSIP) or browser-based softphones.

NAT Traversal & Media Handling Modules

When endpoints sit behind firewalls or CGNAT, these modules help SIP signaling and media flow correctly, even in complex topologies.

nathelper – NAT Detection

nathelper detects and adjusts SIP signaling for endpoints behind NAT. It rewrites Contact headers, inserts proper RTP IPs, and helps maintain registration during NAT refresh cycles. It’s lightweight and effective when paired with an RTP proxy.
When to use it: For SIP clients behind NAT, such as mobile or residential users.

rtpproxy – Media Relay Integration

This module interfaces with an external rtpproxy daemon to relay media between endpoints when direct audio isn’t possible. It’s widely adopted, easy to configure, and well-maintained.
When to use it: To support symmetric NAT traversal, RTP relay, or scenarios requiring media anchoring.

mediaproxy – Advanced Media Proxy

mediaproxy is a more feature-rich RTP relay with policy control, accounting, and tighter integration options. However, it’s heavier and less commonly used due to deployment complexity.
When to use it: For advanced policy control or environments needing detailed media enforcement.

stun – NAT Discovery Support

The stun module provides STUN capabilities to help endpoints determine their public-facing IP address. It’s useful for simpler NAT scenarios where full media proxying isn’t needed.
When to use it: In mobile, peer-to-peer, or lightweight NAT environments without media relays.

Presence, Messaging, and Advanced Features

These modules extend OpenSIPS beyond basic call control, supporting features like BLF (Busy Lamp Field), chat, and multi-modal communication.

presence, pua, xcap_client – Presence Framework

These modules allow OpenSIPS to publish and subscribe to user presence status, such as “available” or “on call.” XCAP support allows external systems to manage presence state over HTTP.
When to use it: For BLF, desktop phone presence, or UC integrations.

imc – In-Message Chat

The imc module enables instant messaging chat between SIP users using SIP MESSAGE methods. It works with SIMPLE-compliant clients and is useful in team communication setups.
When to use it: For chat or messaging in SIP-based collaboration platforms.

b2b_logic – Back-to-Back UA Engine

This powerful module lets OpenSIPS act as a full B2BUA, splitting SIP dialogs and controlling each leg independently. It enables full call isolation, header manipulation, and advanced session logic.
When to use it: For anonymized call routing, call forking, or regulatory separation between call legs.

Scenario-Based Module Recommendations

Not every system needs every module. These are the module sets our OpenSIPS VoIP developers actually use in production, tailored for specific roles like PBX, Class 4 routing, or secure SBCs.

Hosted PBX

To manage endpoint registrations, NAT traversal, multi-device users, and presence features, a hosted PBX stack needs flexibility and resilience across user-facing modules.

• registrar
• usrloc
• dialog
• nathelper
• tls
• permissions
• presence

Class 4 Routing

Carrier-grade SIP routing in Class 4 switches requires high-performance call distribution, dynamic route control, and proper accounting for billing and analytics.

• dispatcher
• drouting
• tm
• acc
• auth
• topology_hiding

Secure SBC

A secure SBC (Session Border Controller) must handle encrypted SIP signaling, WebRTC clients, session isolation, and SIP-level security and monitoring.

• tls
• proto_wss
• auth_db
• permissions
• siptrace
• topology_hiding

How to Choose the Right OpenSIPS Modules?

Choosing the right modules is all about what your architecture needs. This part will help you avoid overloading your config with unused features or missing critical dependencies.

1. Start from your role

• Registrar? You’ll need registrar, usrloc, auth_db, nathelper, tls.
• SBC? Focus on tls, topology_hiding, permissions, proto_wss, and optionally b2b_logic.
• Router? Use dispatcher, drouting, tm, acc.

2. Think about media

Will this node see RTP traffic?

• Yes? You’ll need rtpproxy or mediaproxy.
• No? Focus only on SIP modules and skip media dependencies.

3. Consider transport protocols

• UDP only? Keep it light.
• TLS/WebRTC? You’ll need tls, proto_wss, and proper cert management.

4. Plan for security and compliance

Don’t skip:

• auth + auth_db for SIP authentication
• permissions to block unknown IPs
• topology_hiding to protect internal headers
• acc or dialog for call accounting and audits

5. Know the dependencies

Many modules need others:

• auth_db needs a DB backend like db_mysql
• Presence relies on usrloc and database support
• drouting needs a properly structured DB schema

6. Don’t over-engineer

More modules = more complexity. 

Only enable what you can test, monitor, and maintain. If in doubt, start small and scale as needed.

Conclusion

OpenSIPS modules are what make it powerful. But what makes a VoIP system great is knowing exactly which modules to use and why.

Start with the problem you’re solving, then select only the modules that fit. And if you’re unsure what fits best, don’t gamble with your stack.

Hire developers who’ve already solved these problems for Class 4 carriers, UCaaS platforms, and enterprise VoIP providers!