sip

Lampline

+372

Build AI phone agents on your own computer and test every change with an AI caller. Open source, runs free with no API keys, cents per call with your own keys. Imports Dialogflow CX.

ai-native-callcenter

+144

Self-hosted AI call center: AI talks, code controls the call. AI agents answer every call; humans take over without dropping it. Runs on FreeSWITCH with your SIP trunk and realtime model (OpenAI, Grok, Qwen, Doubao, or any compatible gateway). API-first, with a built-in agent desktop.

freesbc

+43

One binary, one YAML file. A pure-Go Session Border Controller that keeps FreeSWITCH or Asterisk off the public internet: SIP proxy, RTP/SRTP relay, WebRTC gateway, built-in SIP security.

freeswitch

+20

FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a Raspberry PI to a multi-core server, FreeSWITCH can unlock the telecommunications potential of any device.

asterisk

+13

The official Asterisk Project repository.

mirotalksfu

+10

🏆 Self-hosted, open-source WebRTC video conferencing platform for real-time communication and collaboration. A modern alternative to Zoom, Google Meet, Jitsi Meet, and Microsoft Teams, powered by scalable SFU architecture with Mediasoup.

kamailio

+9

Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms, focusing on flexibility, security and scalability

baresip

+5

Baresip is a modular SIP User-Agent with audio and video support

linphone-android

+4

Linphone.org mirror for linphone-android (https://gitlab.linphone.org/BC/public/linphone-android)