OSI-licensed open source

Best Open-Source SIP Servers (2026)

A SIP server handles Session Initiation Protocol signalling for voice and video calls: registering endpoints, routing calls and enforcing policy. Open-source projects fall into two groups, SIP proxies and registrars that route large volumes of signalling, and full PBX or media servers that also handle audio, voicemail and conferencing.

This page covers both groups so you can match the tool to the job. Kamailio and OpenSIPS are scalable SIP proxies, Asterisk and FreeSWITCH are media-capable PBX platforms, Yate is a combined telephony engine, drachtio is a SIP application framework and FreePBX is a GUI on top of Asterisk. Licences were read from each repository.

Tools, licences and pricing checked against vendor and repository pages on November 7, 2026.

What to look for

  • •Role: signalling proxy and registrar versus media and PBX server
  • •Scale and performance under many concurrent registrations or calls
  • •Protocol support: WebRTC, TLS, SRTP, IPv6 and transcoding
  • •Configuration model: scripting language, GUI or APIs
  • •Community, documentation and commercial support options

At a glance

ToolBest forPricingLicence
AsteriskSelf-hosted PBX and telephony applicationsFree, open source (commercial licensing and support available from Sangoma)GPL-2.0
FreeSWITCHScalable media handling and softswitch buildsFree, open sourceMPL-1.1
KamailioHigh-volume SIP proxying and load balancingFree, open sourceGPL-2.0
OpenSIPSCarrier-grade SIP routing and VoIP platformsFree, open sourceGPL-2.0
YateProtocol-bridging telephony engineFree, open sourceGPL-2.0
drachtioDevelopers building SIP apps in Node.jsFree, open sourceMIT
FreePBXTeams wanting a GUI-managed Asterisk PBXFree, open source (paid commercial modules available from Sangoma)GPL-3.0

1. Asterisk

GPL-2.0

Best for: Self-hosted PBX and telephony applications · Free, open source (commercial licensing and support available from Sangoma)

Asterisk is an open-source framework for building communications applications, widely used as a PBX. It supports SIP, voicemail, IVR, conferencing and many integrations.

Strengths

  • + Huge community and documentation
  • + Flexible dial plan and APIs

Trade-offs

  • − Single-process design limits massive scale
  • − Configuration has a learning curve

2. FreeSWITCH

MPL-1.1

Best for: Scalable media handling and softswitch builds · Free, open source

FreeSWITCH is a telephony platform for voice, video and messaging that is used for softswitches, conferencing and WebRTC gateways. It exposes event socket and scripting interfaces.

Strengths

  • + Strong media and conferencing capabilities
  • + Scripting and event APIs

Trade-offs

  • − Complex configuration
  • − Project governance has shifted over time

3. Kamailio

GPL-2.0

Best for: High-volume SIP proxying and load balancing · Free, open source

Kamailio is an open-source SIP server used as a proxy, registrar, load balancer and routing engine. It is known for handling very high volumes of SIP signalling with a flexible routing script.

Strengths

  • + Very scalable signalling
  • + Flexible routing scripts and modules

Trade-offs

  • − No media handling by itself
  • − Needs SIP expertise

4. OpenSIPS

GPL-2.0

Best for: Carrier-grade SIP routing and VoIP platforms · Free, open source

OpenSIPS is a multi-functional SIP server for routing, registration, load balancing and VoIP service platforms. It provides a scripting language and many modules.

Strengths

  • + High performance signalling
  • + Strong module and management interface ecosystem

Trade-offs

  • − No built-in media server
  • − Steep learning curve

5. Yate

GPL-2.0

Best for: Protocol-bridging telephony engine · Free, open source

Yate (Yet Another Telephony Engine) is a telephony engine that handles SIP, H.323 and other protocols and can act as a PBX or gateway. Its modules are scriptable with languages like PHP and Python.

Strengths

  • + Multi-protocol support
  • + Scriptable modules

Trade-offs

  • − Smaller community
  • − Documentation is thinner than Asterisk

6. drachtio

MIT

Best for: Developers building SIP apps in Node.js · Free, open source

drachtio is an open-source SIP server framework that lets Node.js applications handle SIP signalling. It is commonly paired with a media server such as FreeSWITCH for voice apps.

Strengths

  • + Developer-friendly Node.js API
  • + Works with external media servers

Trade-offs

  • − Not a ready-made PBX
  • − Requires separate media server

7. FreePBX

GPL-3.0

Best for: Teams wanting a GUI-managed Asterisk PBX · Free, open source (paid commercial modules available from Sangoma)

FreePBX is a web-based graphical interface for managing Asterisk. It handles extensions, trunks, call routing and voicemail without hand-editing configuration files.

Strengths

  • + Approachable GUI
  • + Large user base

Trade-offs

  • − Some modules are commercial
  • − Tied to Asterisk

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How to choose

If you need…Start with
Large-scale SIP routingKamailio, OpenSIPS
All-in-one PBXAsterisk, FreePBX
Conferencing and media at scaleFreeSWITCH
Node.js voice appsdrachtio
Protocol bridgingYate

Best Open-Source SIP Servers: FAQ

Quick answers to common questions

What is the difference between a SIP proxy and a PBX?

A SIP proxy routes signalling but does not touch audio. A PBX or media server also handles call media, voicemail, IVR and conferencing.

Kamailio or OpenSIPS?

Both are mature SIP proxies. Choose based on module needs, team familiarity and the management tooling you want.

Can I use Kamailio and FreeSWITCH together?

Yes. A common design puts Kamailio or OpenSIPS in front for routing and load balancing, with FreeSWITCH or Asterisk behind for media.

Is Asterisk free for commercial use?

Asterisk is available under GPL-2.0, and alternative commercial licences are negotiated with Sangoma. Check the GPL obligations for your use.

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