VoIP en telefonie

Open source voice AI platform. Self-hosted alternative to Vapi and Retell. On Prem, BYOK across Speech to Speech or LLM/STT/TTS, with a visual workflow builder, MCP native and telephony support.

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.

PJSIP project

Complete, modular and scalable SIP server, includes a push gateway, to deliver SIP incoming calls or text messages on mobile device platforms where push notifications are required to receive information when the app is not active in the foreground.

Modular SIP server (registrar/proxy/router/etc).

Low-latency, high quality voice/text chat software.

Async PHP client API for the telegram MTProto protocol

Linphone is a free VoIP and video softphone based on the SIP protocol. Mirror of git://git.linphone.org/linphone-desktop.git

Modern and flexible SIP/VoIP cli tool

Troubleshooting and monitoring VoIP calls.

A complete voice AI building block with telephony integration, featuring real-time speech-to-text, text-to-speech, and LLM-powered conversational agents.

JUICE is a UDP Interactive Connectivity Establishment library

Open-source, self-hosted AI phone receptionist powered by OpenAI Realtime API. High-fidelity speech-to-speech voice agent. Drop-in replacement for overpriced AI receptionist SaaS.

Self-hosted, API-first AI call center. Realtime voice bots answer every call, human agents take over without a drop. Full REST + SSE API with OpenAPI spec to embed in your product. FreeSWITCH + OpenAI Realtime / Qwen, one Go binary.

🚀 The open-source alternative to Twilio.

Modern XMPP ("Jabber") Chat Client using GTK/Vala

XMPP client for macOS based on TigaseSwift XMPP library

Lightweight SIP proxy, location server, and registrar for a reliable and scalable SIP infrastructure.

SIP proxy/server for voice, video, IM, presence and any other SIP extensions.

Set of tools to audit SIP based VoIP Systems

Pure Go implementation of the WebRTC API

E2E test framework for tests with complex environment requirements.

An open source Session Border Controller 🌟 The SBC you dream about 🗽 LibreSBC will help you save thousands of dollars.

Ncurses SIP Messages flow viewer

✨📞 Asterisk PBX in 🐳 Docker — Smallest Asterisk ever! 🚀

SIP Security Assessment Framework for VoIP Pentesters. Presented at DEFCON, BlackHat & Offzone.

Async client API for the Telegram Calls

100% Open-Source Packet Capture Agent for HEP

A SIP client inside home assistant!

Easy to use but advanced IP PBX system, VoIP gateway and conference server.

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

Web interface for multi-platform voice switch called FreeSWITCH.

A Go implementation of STUN

HEP Capture Server for HOMER

Self-hosted native-Rust runtime for real-time voice agents. Own the stack: one binary in your own VPC or air-gapped, no hosted control plane. pipecat-compatible pipeline, in-process SIP/RTP, single-process call density. Apache-2.0.

Scalable open source cross-platform telephony platform.

baresip library based SIP client for Android

Home-Assistant SIP Gateway

A PBX written by rust

Linphone is a free VoIP and video softphone based on the SIP protocol. Mirror of linphone-iphone (git://git.linphone.org/linphone-iphone.git)

Web Base Telnyx/Twilio VoIP Interface enabling SMS, MMS and Voice calls using the providers API Keys and self-hosting the application.

HOMER 7.x Front-End and API Server

IVOZ Provider - Multitenant solution for VoIP telephony providers

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

Twilio C#/.NET Helper Library for .NET6+.

Decentralized, Global-Scale Media Server written in Rust (WebRTC/Whip/Whep/Rtmp/Sip)

Portable and Lightweight HEP Capture Agent for HOMER

P-KISS-SBC - simple and stupid SIP/RTP SBC - AGPL v3 - Based on kamailio / RTP Engine

WebRTC plugin for Flutter Mobile/Desktop/Web

A Java library for communicating with the Twilio REST API and generating TwiML.

A dart-lang version of the SIP UA stack.

Voglander 是基于 Spring Boot 3.5 + Java 17 的企业级视频监控平台，支持 GB28181、JT/T 1078、ONVIF、SIP Gateway、ZLMediaKit、IPC/NVR 接入、实时点播、录像回放、流媒体转发、NAT 穿透和国标级联。

Modern open-source multi-tenant PBX built on FreeSWITCH, Laravel and Vue. An alternative to FusionPBX, FreePBX and 3CX.

GNU Gatekeeper - H.323 server for VoIP and videoconferencing

Open-source MCP server for safe, stateful AI-agent phone calls with OpenAI Realtime and Twilio.

一个使用 Rust 构建的智能视音频监控平台，从设备接入、媒体处理、流媒体分发与视觉分析，到灵活部署 边缘端、中心端、云端，完成端到端交付

A Native Implementation of Telegram Calls in a seamless way.

Web-based open source GUI that controls and manages Asterisk.

Full-featured IPBX solution built atop Asterisk with integrated Web administration interface and REST-ful API.

Transit class4 softswitch(SBC) with integrated billing and routing engine and REST API.

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

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.

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 Go implementation of the SDP

Android app for Mumble voice conference system. — NOTE WELL that htis is a mirror of https://gitlab.com/quite/mumla — Please file issues and merge requests over there. (Mirror of the Humla library here: https://github.com/quite/humla)

Simlar for android

Self-contained proximity voice chat solution for Unity with easy setup and support for many networking libraries. Not affiliated with Meta Platforms, Inc!

SIP Stack Rust library for building SIP applications

RODA - Repository of Authentic Digital Objects

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

API server and Web GUI for FreeSwitch written in Golang and Angular

The RTPproxy is a high-performance software proxy for RTP streams that can work together with Sippy B2BUA, Kamailio, OpenSIPS and SER.

🤙Learning VoIP, RTP, pjsip and SIP

Minecraft + Mumble + MumbleLink = You hear where voices come from and how far away they are! This is a Minecraft mod based on "Minecraft Forge". It's purpose is to be able to use Minecraft in conjunction with Mumble's positional audio feature (http://mumble.sourceforge.net/).

MagnusBilling is a fast, secure, efficient, high availability, VOIP Billing.

SIP in Go

A plugin that allows the flutter app to communicate with a Janus server using different transport mechanisms, such as WebSocket and HTTP(rest). It features a developer-friendly api to interact with various WebRTC Janus Plugins. Hence, it can be considered as a swiss-knife for WebRTC solutions.

Turn any SIP call into a realtime AI voice agent (OpenAI Realtime / Deepgram/Gemini Live/xAI Grok Voice)

Modern Android dialer with built-in contacts, fast call history, AMOLED themes, dual-SIM support, privacy-focused calling, and rich contact management — a powerful alternative to Google Phone

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

SIP softphone for Mac

纯协议层框架，屏蔽 SIP 协议细节。业务方实现 Listener 接口（推荐）或监听 Layer 1 协议事件 接收消息，通过 ClientCommandSender / ServerCommandSender 发送命令，不直接接触 JAIN-SIP。框架内置 GB28181-2016 + GB/T 28181-2022 协议实现，单 JVM 可同时启用平台服务端（gb28181-server）和设备客户端（gb28181-client），支持级联代理场景。

Public Project of Tigase Messenger for iOS devices based on Tigase Swift XMPP library.

A GoLang FreeSWITCH ESL Library

The world's only Open Source CPaaS with a Twilio compatible API

Dial Plan Vizualizer module for FreePBX

Open-source eBPF sensor for IPv4/UDP SIP and correlated RTP/RTCP metrics in Prometheus and Grafana. Support: https://github.com/aibudaevv/sip-exporter/issues

即时通讯(IM)系统

🚨 Home Security Intrusion Detection for Raspberry PI or any other linux OS

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

Low latency, web based, open source, high quality, voice-over-ip intercom solution

BTicino Classe 300X13E and C100X

AI Phone Agent: A starter kit to build AI agents that answer real phone calls and talk to customers in real time. Amazon Connect, OpenAI SIP, and the OpenAI Realtime API. Dial +1 844-593-8442 to try the demo.

Advanced repeater system software with EchoLink support for Linux including a GUI, Qtel - the Qt EchoLink client

Voice chat/VoIP solution for unity.

Docker image providing Asterisk PBX

Open Source Fax Server Software supporting inbound and outbound faxing. ICTFAX can be used in following faxing scenarios Web to fax, Fax to Web or Online Fax Email to fax and fax to email ATA / Extensions to send and receive faxes via Fax machine using internet REST based Fax API’s to integrate with third party applications and automate business process G.711 based Fax Origination / Termination / Gateway T.38 based Fax Origination / Termination PSTN based Fax Origination / Termination / Gateway

Let your AI make real phone calls from your own phone . Cellular call gateway for Android, Windows, Linux, Hermes, and OpenClaw.

NetToggle brings advanced cellular network controls to Android Quick Settings. Switch between 5G Only (SA/NR), 4G/LTE Only, Preferred 5G, Preferred 4G, Preferred 3G, and 2G Only. Build custom tile cycles, target one or both SIMs, create shortcuts, use secure broadcast automation, restore preferred modes, and diagnose failures using Root or Shizuku.

Dockerized FreePBX 17 with Asterisk 21, PHP 8,2, dedicated MariaDB and data persistence

baresip python wrapper

SIP softphone

FlashVAD is a small causal voice-activity detector and integration toolkit for low-latency voice calls. It emits one speech probability every 10 ms

A collection of exploits for different VoIP products.

SIP3 Captain (Community Edition)

Voip Open Linear Testing Suite

a Python extension that provides bindings to WebRTC M92

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.

VoIP troubleshooting and monitoring platform.

Pàdé (Yoruba word for Meet) is a browser extension (Chrome/Edge) based unified communications desktop client for Openfire.

WebSocket-based API and SDKs to connect to Zello channels (BETA)

SIP3 Salto (Community Edition)

EZK is a collection of Rust crates to build SIP applications

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

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.

A SIP call processing server that can be controlled via nodejs applications

JavaForce library for building powerful Apps and Services (Native Bindings for Camera, OpenGL, Vulkan, OpenCL, ffmpeg). Includes VoIP stack, PLC I/O and many apps.

Entrypoint for the Somleng Project

SIP3 Twig (Community Edition)

Video Conferencing application with support for P2P Mesh - SFU hybrid video conferencing

SomlengSWITCH. Somleng's TwiML Engine, FreeSWITCH configuration and infrastructure.

FreeSWITCH module that streams live call audio to AI voice agents over WebSocket and plays their voice back — full-duplex.

TeamTalk 5 Development

Popular Android messaging app for VoIP.ms, a Canadian VoIP provider

Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://inextrix.atlassian.net/jira/projects

SMS, Fax, Voice Broadcasting and auto dialer Software, A unified communications open source autodialer developed over freeswitch communication engine and ICTCore communications framework

A utility to automate the installation, maintenance, and debugging of Asterisk/DAHDI, while integrating additional patches to provide the richest telephony experience

Ansible scripts to install and configure SIP3

Rust implementation of a mumble server for FiveM

SIP3 Commons

OpenSource Freeswitch & Kamailio Billing, rating and Routing Software

iOS CallKit framework and Android ConnectionService for React Native

SIP视频对讲、MCU融屏、指挥调度 、网页视频会议、视频会议、H5视频直播、SIP广播会议、融合通信、WebRtc、RTMP、SIP、RTSP、白板、视频录制、视频通话、旁路直播、AR眼镜、执法记录仪、AR专家系统、AR远程协助、AR巡检、远程医疗、视频安全帽、机顶盒、会议转录、ASR/TTS

即时通讯，聊天，野火IM Android客户端，支持Android 4.x —— 最新

A simple, intuitive, and powerful JavaScript signaling library

The Open Source Contact Center Solution (mirror of https://gitlab.com/omnileads/ominicontacto)

📞 CallKit + Core-Telecom for React Native + Expo. A modern react-native-callkeep alternative.

mjSIP - a complete Java-based SIP stack implementation

Flutter Twilio Voice Plugin

Flutter SDK for CallKit integration to Flutter applications on iOS

Protect your SIP Servers from bad actors at https://sentrypeer.org

Simple, friendly and desktop integrated VoIP client.

HOMER 7 Docker Images

SIPVicious OSS is a VoIP security testing toolset. It helps security teams, QA and developers test SIP-based VoIP systems and applications. This toolset is useful in simulating VoIP hacking attacks against PBX systems especially through identification, scanning, extension enumeration and password cracking.

A SIP/WebRTC voice agent

Easier web calling by providing a layer of abstraction around SIP.js

VoIP signaling and media test automation

VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine

:telephone_receiver: Open Source Telephony API Platform

An automated testing framework for Interactive Voice Response (IVR) call flows

Full-featured PABX system, with call queues, built-in web conferencing, live chat and social media messaging all on one system.

Showcase app demonstrating Waterbus-powered real-time media.

智能电话外呼系统 呼叫中心系统 freeswitch webrtc

Barebone Opensource Powered SBC

Open-source event-driven AI powered Softphone

Call Center，Contact Center，AI，呼叫中心，客服系统，工单系统，智能外呼，大模型呼叫中心，智能呼叫中心，FreeSWITCH

H323Plus - H.323 development framework

FreeSWITCH externals projects

A fully featured browser based WebRTC SIP phone for Asterisk

AI-powered SIP PCAP analyzer with call flow, hold/resume detection, and GPT-4o chat

FreeSwitch library for DotNet

The JavaScript framework to leverages the power of Kazoo

Flutter Video SDK - Build your own video app experience using Dart, Flutter and the Stream Video Messaging API.

Yeti admin interface. Opensource SIP SBC and VOIP billing.

Rocket Connect - An agnostic connector for Chat Platforms

T-one is a high-performance streaming ASR pipeline for Russian, specialized for the telephony domain.

an easy-to-use, secure and decentralized conference call application (this repository is a mirror of an internal work repository)

SIP swiss army knife

📱 Android app providing ⚡ REST API for call management - companion to SMSGate app 🔀

Pure python VoIP/SIP/RTP library. Currently supports PCMA, PCMU, and telephone-event

Voice chats, private incoming and outgoing calls in Telegram for Developers

SIPp for Humans - launch multiple agents with Python

Minimalist Windows / OSx / Linux SIP Softphone with API Control

Battle proven FreeSWITCH Event Socket Protocol client implementation with Gevent

bulk outbound calls with automatic Whisper transcription via Telnyx

SIP3 Parent POM

An introduction to the 46elks API with code examples

Pluggable WebRTC softphone and communication platform.

Unity Voice Chat Asset

A PJSIP module for React Native.

:iphone: [Android Library] Get device information in a super easy way.

OpenTok demo application

:books: WebRTC (Web Real-Time Communications) 中文教程

An all-in-one Lsposed module to spoof all things networking (including eSIM compatibility)!

asyncio powered FreeSWITCH cluster control

godot-voip is a Godot Engine addon which makes it very easy to setup a real-time voice-chat system in your Godot game. This addon also includes a demo project.

:trophy: Feature packed, easy-to-use game server API for Go back-ends and Javascript clients. Tutorials and examples included!

A Fork from http://git.freeswitch.org/git/freeswitch-contrib/tree/dvarnes/java/esl-client

SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. SaraPhone is fully integrated with FusionPBX. Based on SIP.js, SaraPhone works with all WebRTC compliant servers: FreeSWITCH, Asterisk, OpenSIPS, Kamailio, etc. SaraPhone gets its name from Giovanni's wife, Sara.

WireBug is a toolset for Voice-over-IP penetration testing

PHP SIP User Agent class

☎️ Original open source call flooder using Twilio's API.

The ITU-T Software Tool Library (G.191)

Simple and fast viewer for Asterisk CDRs and Recordings (Mod)

Go SIP UA library for client/b2bua

WebCall for Android - Web-Telephony P2P Messaging File-Exchange E2E-Encryption No-SIM

一对一呼叫、邀请呼叫、音视频通话、多人通话，适合陌生人交友、在线教学、在线医疗、智能终端等场景；更能轻松实现腾讯会议，Zoom会议，钉钉课堂等多人音视频交互类场景。一对一音视频，一对多音视频。

SIP support for AsyncIO (DEPRECATED)

Android Mobile SDK to easily integrate communication features (WebRTC, messaging, presence, voice, video, screensharing) based on RestComm into native Mobile Applications

libice是一个c/c++库，它实现了RFC5245规范定义的交互式连接建立(ICE)协议, 适用于实时通信领域，比如音视频实时通信。