Self-hosted VoIP / SIP & PBX tools
14 open source alternatives in this category
See also:
Calendars & Contacts (CalDAV/CardDAV)Collaborative Office SuitesEmail Servers (SMTP/IMAP)Team Chat & MessagingVideo Conferencing & WebRTCWebmail Clients14 services found
Mumble
Low-latency, high-quality VoIP voice chat client and server
Mumble is an open-source, low-latency VoIP voice chat platform with a desktop client and a self-hosted server for encrypted group communication.
ejabberd
Scalable real-time messaging server for XMPP, MQTT, and SIP
ejabberd is an Erlang/OTP-based messaging server providing XMPP chat and presence, MQTT broker capabilities for IoT, and SIP services for real-time communications.
FreeSWITCH
Software-defined telecom stack for VoIP, SIP, and real-time media
FreeSWITCH is an open-source software-defined telecom stack for building VoIP and real-time communication services with SIP and WebRTC support.
Asterisk
Open-source telephony engine and PBX framework
Open-source PBX and telephony toolkit for building communications applications; modular C-based engine with SIP, WebRTC, RTP, ARI/AMI APIs and hardware support.
Kamailio
High-performance SIP server for VoIP and real-time communications
Kamailio is an open source SIP signaling server for scalable VoIP and real-time communication platforms, supporting routing, load balancing, WebRTC, and IMS/VoLTE.
HOMER (SIPCAPTURE)
Packet capture and observability for SIP, VoIP, and WebRTC
Carrier-grade SIP/VoIP/WebRTC packet capture and monitoring platform with real-time search, correlation, and troubleshooting workflows using the HEP/EEP protocol.
Routr
Programmable SIP proxy, registrar, and location server
Routr is a programmable SIP server (proxy/registrar/location) for building scalable VoIP infrastructure with APIs, multi-tenancy, and Kubernetes-ready deployment.
FusionPBX
Multi-tenant PBX and softswitch web interface for FreeSWITCH
FusionPBX is a domain-based, multi-tenant PBX and softswitch web UI for FreeSWITCH, supporting extensions, IVR, queues, voicemail, provisioning, and more.
Tigase XMPP Server
Modular, high-performance XMPP/Jabber server in Java
Scalable, modular XMPP/Jabber server written in Java supporting TCP, BOSH, WebSockets, federation, components, HTTP API and push notifications.
Flexisip
Modular C++ SIP server suite for VoIP, messaging, and conferencing
Open-source, modular SIP server (proxy, presence, conference, B2BUA) with push gateway, ICE/STUN/TURN, MySQL/SQLite support; optimized for mobile and embedded systems.
FreePBX
Web-based open-source GUI for managing Asterisk-based IP PBX systems
Modular, PHP/JavaScript web GUI that configures and manages Asterisk PBX features, endpoints, and call routing for businesses and service providers.
SIP3
SIP and VoIP QoS monitoring with call flow analytics
Self-hosted platform for capturing SIP/RTP traffic, analyzing call flows, and monitoring VoIP quality metrics with dashboards and alerts.
Wazo Platform
Open-source programmable IP telecommunication platform
Open-source, API-first platform for carrier-grade IP communications: VoIP, WebRTC, messaging, conferencing and programmable telephony microservices.
Yeti-Switch
Carrier-grade Class 4/5 VoIP softswitch and SBC
Carrier-grade VoIP softswitch/SBC for routing, billing, and managing SIP traffic, with LCR, fraud control, and high-availability deployment options.