Flexisip

Flexisip

Modular C++ SIP server suite for VoIP, messaging, and conferencing

174stars
78forks
Last commit: 1mo ago
Repo age: 11y old

Flexisip is a modular, scalable SIP server suite written in modern C++. It provides proxying, presence, conferencing, push gateway and B2BUA capabilities to build VoIP, messaging and real-time communication services.

Key Features

  • Modular server suite: proxy, presence server, conference server, B2BUA and an account manager as separate components.
  • Push Gateway: routes SIP notifications to mobile platforms and supports RFC 8599 and current APNs/Firebase requirements.
  • Media and NAT handling: built-in media relay with ICE/STUN/TURN support and RTP/RTCP/SRTP handling for NAT traversal and secure media.
  • Authentication and security: Digest, TLS client certificate authentication, TLS (OpenSSL) and optional OpenID Connect/OAuth support.
  • Account management and remote config: REST API (FlexiAPI) for user/account lifecycle, dynamic per-user configuration, and a web admin platform for service management.
  • Scalability and deployment: designed for high-availability cluster deployments, load balancing and low-footprint embedded targets (Raspberry Pi, IoT).
  • Interoperability: compatible with other SIP systems and extensions; supports SIP transport over UDP/TCP/TLS and various SIP RFC extensions.

Use Cases

  • Deploy a hosted SIP service with calling, messaging and conferencing features for mobile and desktop clients.
  • Add push-notification support in front of legacy SIP platforms to deliver calls/messages reliably to smartphones.
  • Embed a lightweight SIP proxy in edge/IoT devices (intercoms, access control) to add audio/video calling features.

Limitations and Considerations

  • PSTN interconnection requires a separate trunk or gateway; Flexisip provides B2BUA functionality but does not supply PSTN lines or virtual numbers.
  • Some optional features require additional native dependencies (for example MySQL for account DB clustering, Redis for registrar/cluster communication, XercesC for presence features), so build-time configuration impacts feature set.
  • Dual-licensing (AGPLv3 or proprietary) may impose AGPL obligations for some deployments; evaluate licensing before commercial use.

Flexisip combines a full set of SIP server components with mobile-friendly features and small-footprint deployment options. It is targeted at teams building unified communication services, embedded SIP-enabled devices, and operators needing push-enabled SIP gateways.

Categories:

Tags:

Tech Stack:

Share:

Similar Services

Mumble

Mumble

Low-latency, high-quality VoIP voice chat client and server

7.5k
1.3k
Last commit: 22h ago

Mumble is an open-source, low-latency VoIP voice chat platform with a desktop client and a self-hosted server for encrypted group communication.

Alternative to:
Skype
Skype
+4
ejabberd

ejabberd

Scalable real-time messaging server for XMPP, MQTT, and SIP

6.5k
1.5k
Last commit: 1d ago

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.

Alternative to:
CloudMQTT
CloudMQTT
+11
FreeSWITCH

FreeSWITCH

Software-defined telecom stack for VoIP, SIP, and real-time media

4.6k
1.7k
Last commit: 20h ago

FreeSWITCH is an open-source software-defined telecom stack for building VoIP and real-time communication services with SIP and WebRTC support.

Alternative to:
Twilio
Twilio
+15
Asterisk

Asterisk

Open-source telephony engine and PBX framework

3k
1.2k
Last commit: 2d ago

Open-source PBX and telephony toolkit for building communications applications; modular C-based engine with SIP, WebRTC, RTP, ARI/AMI APIs and hardware support.

Alternative to:
3CX
3CX
+19
Kamailio

Kamailio

High-performance SIP server for VoIP and real-time communications

2.7k
1.1k
Last commit: 2d ago

Kamailio is an open source SIP signaling server for scalable VoIP and real-time communication platforms, supporting routing, load balancing, WebRTC, and IMS/VoLTE.

Alternative to:
AudioCodes Mediant Session Border Controller (SBC)
AudioCodes Mediant Session Border Controller (SBC)
+6
HOMER (SIPCAPTURE)

HOMER (SIPCAPTURE)

Packet capture and observability for SIP, VoIP, and WebRTC

1.9k
251
Last commit: 1mo ago

Carrier-grade SIP/VoIP/WebRTC packet capture and monitoring platform with real-time search, correlation, and troubleshooting workflows using the HEP/EEP protocol.

Alternative to:
ThousandEyes VoIP Monitoring
ThousandEyes VoIP Monitoring
+3