sip

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.

Jitsi is an audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, IRC and many other useful features.

RTSP/RTP/RTMP/FLV/HLS/MPEG-TS/MPEG-PS/MPEG-DASH/MP4/fMP4/MKV/WebM

The official Asterisk Project repository.

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

PJSIP project

A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.

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

Rapida is an open-source, end-to-end voice AI orchestration platform for building real-time conversational voice agents with audio streaming, STT, TTS, VAD, multi-channel integration, agent state management, and observability.

Official one-stop shop for AI Agents and developers building with Telnyx.

Open-source realtime voice agent server in Go with WebRTC (WHIP), barge-in, streaming STT/LLM/TTS pipelines, plugin system, multi-language SDKs, SIP telephony, ESP32 support & fully local mode.

A programmable Voice AI platform: SIP and WebRTC call control, multi-party mixing, recording, TTS/STT, and pluggable AI agents (ElevenLabs, VAPI, Pipecat, Deepgram) — all driven through a REST API, webhooks, and a WebSocket event stream

The open-source foundation for production Voice AI. Build, scale, and own your AI calling infrastructure. Bridge legacy SIP to modern LLMs over ultra-low latency WebRTC pipelines. Zero platform fees, BYOK, and full data sovereignty.

openduplex uses speech-to-text, artificial intelligence and text-to-speech, to call businesses and make appointments for you

Full-stack LiveKit voice agent starter: a Python voice worker, FastAPI token server, and React frontend. Run, deploy, and swap each piece on its own. Ask for ShipVoice pro

This project is the backend engine for a fully autonomous AI-powered call center. It integrates a large language model (LLM), speech recognition, and text-to-speech to manage real-time phone conversations via Asterisk.

Open-source SIPREC Session Recording Server (SRS) written in Go — RFC 7865/7866 compliant, multi-vendor SBC support, AI-powered speech-to-text, and horizontally scalable

Let your AI agent call your phone and talk to you — MCP servers for live, interruptible voice calls + tiered alerts, using free self-hosted pieces (pjsua2 + whisper.cpp + Linphone). No paid telephony, no extra API key.