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.
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.
Pure Go implementation of the WebRTC API
A free and open source instant messaging and VoIP chat app built for friends, groups, and communities.
Async PHP client API for the telegram MTProto protocol
Java wrapper for the popular chat & VOIP service: Discord https://discord.com
DistroAV (formerly OBS-NDI): NDI integration for OBS Studio
The official Asterisk Project repository.
MikoPBX — free open-source PBX for small business, built on Asterisk 22, PHP 8.4, and Phalcon 5. Deploy as ISO, Docker, LXC, or in the cloud.
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.
Open-source workspace for teams: chat, voice & video calls, task boards, wiki, file drive, calendar and even emails — in one app. Self-hosted, fast on any computer or phone.
FreeSWITCH module that streams live call audio to AI voice agents over WebSocket and plays their voice back — full-duplex.
Robust, Ubiquitous and Massively Scalable Messaging Platform (XMPP, MQTT, SIP Server)
Mumble is an open-source, low-latency, high quality voice chat software.
The AI that answers the phone.
WebRTC plugin for Flutter Mobile/Desktop/Web
Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms, focusing on flexibility, security and scalability
即时通讯,聊天,野火IM Android客户端,支持Android 4.x —— 最新
PJSIP project
HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
This TeamSpeak 6 Server Beta repository contains the latest beta version of the TeamSpeak 6 Server software, offering early access to improvements and bug fixes. This beta release is intended for testing and feedback before the stable version launch.
Whatomate is an open-source WhatsApp integration
⚡ The future of programmable SIP servers.
Linphone.org mirror for linphone-android (https://gitlab.linphone.org/BC/public/linphone-android)
即时通讯,聊天,野火IM iOS客户端
SIP softphone for Mac
iOS CallKit framework and Android ConnectionService for React Native
Group calls powered by Matrix
Element Server Suite Community Edition
A Go implementation of STUN
Go implementation of the MediaDevices API.
Advanced repeater system software with EchoLink support for Linux including a GUI, Qtel - the Qt EchoLink client
Linphone is a free VoIP and video softphone based on the SIP protocol. Mirror of git://git.linphone.org/linphone-desktop.git
Open-source, self-hosted TeamSpeak 3/6 client for the browser — connect to any reachable server.
SIP/RTP signaling & media quality monitor
Open-source IP PBX & unified communications: Asterisk 22 web portal, WebRTC softphone, chat, fax, call center, Android & iOS apps — https://aipbx.bid
Baresip is a modular SIP User-Agent with audio and video support