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
Free, 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.
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.
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.
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
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.
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.
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.