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.

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

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

PJSIP project

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.

Async PHP client API for the telegram MTProto protocol

🚀 The open-source alternative to Twilio.

Troubleshooting and monitoring VoIP calls.

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

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

JUICE is a UDP Interactive Connectivity Establishment library

Modern and flexible SIP/VoIP cli tool

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

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.

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

Pure Go implementation of the WebRTC API

E2E test framework for tests with complex environment requirements.

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

Ncurses SIP Messages flow viewer

Set of tools to audit SIP based VoIP Systems

Scalable open source cross-platform telephony platform.

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)

A Go implementation of STUN

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

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

Web interface for multi-platform voice switch called FreeSWITCH.

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

XMPP client for macOS based on TigaseSwift XMPP library

A PBX written by rust

Async client API for the Telegram Calls

WebRTC plugin for Flutter Mobile/Desktop/Web

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

A SIP client inside home assistant!

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

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

baresip library based SIP client for Android

Home-Assistant SIP Gateway

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.

HEP Capture Server for HOMER

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

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

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

Portable and Lightweight HEP Capture Agent for HOMER

100% Open-Source Packet Capture Agent for HEP

GNU Gatekeeper - H.323 server for VoIP and videoconferencing

A dart-lang version of the SIP UA stack.

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.

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。

IVOZ Provider - Multitenant solution for VoIP telephony providers

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

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

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

:satellite: A curated list of awesome Real Time Communications resources

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

即时通讯(IM)系统

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

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

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

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

SIP in Go

🤙Learning VoIP, RTP, pjsip and SIP

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.

SIP Stack Rust library for building SIP applications

A Go implementation of the SDP

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

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

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)

VoIP troubleshooting and monitoring platform.

Voice chat/VoIP solution for unity.

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

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 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.

A GoLang FreeSWITCH ESL Library

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

Docker image providing Asterisk PBX

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

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.

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

A collection of exploits for different VoIP products.

BTicino Classe 300X13E and C100X

A simple, intuitive, and powerful JavaScript signaling library

SIP softphone

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

OpenSource Freeswitch & Kamailio Billing, rating and Routing Software

iOS CallKit framework and Android ConnectionService for React Native

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

TeamTalk 5 Development

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

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

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

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

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.

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

智能电话外呼系统 呼叫中心系统 freeswitch webrtc

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

Showcase app demonstrating Waterbus-powered real-time media.

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

A SIP/WebRTC voice agent

H323Plus - H.323 development framework

Yeti admin interface. Opensource SIP SBC and VOIP billing.

A fully featured browser based WebRTC SIP phone for Asterisk

VoIP signaling and media test automation

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

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

FreeSWITCH externals projects

Open-source event-driven AI powered Softphone

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

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

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

Pluggable WebRTC softphone and communication platform.

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

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

SIPp for Humans - launch multiple agents with Python

SIP swiss army knife

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

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

Battle proven FreeSWITCH Event Socket Protocol client implementation with Gevent

A PJSIP module for React Native.

Minimalist Windows / OSx / Linux SIP Softphone with API Control

bulk outbound calls with automatic Whisper transcription via Telnyx

asyncio powered FreeSWITCH cluster control

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

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

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!

Go SIP UA library for client/b2bua

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

a Python extension that provides bindings to WebRTC M92

The Open Source Cloud Communications Platform

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

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。

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

🚀starRTC，即时通讯(IM)系统，免费IM系统（含单聊，群聊，聊天室，文件传输），免费一对一视频聊天，VOIP，语音对讲（回音消除），直播连麦，视频直播，RTSP拉流，RTMP推流，webRTC服务端，在线教育，白板，小班课，在线会议，视频会议，视频监控，局域网直连（无需服务器），兼容webRTC, 支持webRTC加速，P2P高清传输，安卓、iOS、web互通，支持门禁对讲，可视对讲，电视盒子，树莓派，海思，全志，任天堂switch，云游戏，OTT设备，物联网平台，C语言自研方案，支持二次开发成类微信，类映客等APP，✨万水千山总是情，来个star行不行✨，更多示例请访问：

免费IM系统，IM即时通信消息系统(含一对一文字聊天，群聊，聊天室)，免费一对一voip实时通话，录屏，webrtc服务端，免费直播连麦，互动直播，视频直播，RTSP拉流，RTMP推流，语音对讲，免费在线会议，视频会议等服务端程序，支持物联网平台，✨万水千山总是情，来个star行不行✨

SIPp scenarios I use for testing SIP stuff

OpenSource VoIP Door-Intercomstation

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。

Linux/Win/Docker/kubernetes/Chart/Kustomize/GB28181/SIP/RTP/SDP/WebRTC/作为上下级域/平台级联互联

Erlang SIP

Generic library for real-time communications with async IO support

A python-based automated call attendant, call blocker, and voice messaging system running on a Raspberry Pi. Screens callers and block robocalls and scams with a low-cost Raspberry Pi and modem.

sipserver,GB28181,ZLMediaKit

网关服务：Sip与Rtc互通，实现Web，Android，iOS，小程序，SIP座机，PSTN电话，手机互通。

一对一voip视频聊天，直播连麦，多人视频会议，在线会议web演示：

Setup for a WEBRTC client and Kamailio server to call SIP clients

TeamSpeak 3 Client Plugin SDK

Extracting Apple ID Validation Record, Certificate, and Key for AirDrop

RTP Tools

Replacement for the SCCP channel driver in Asterisk. Extended features include Shared Lines, Presence / BLF, customizable Feature Buttons, and Custom Device State. Visit our discussion mailing list for help and join us as a developer if you like.

基于Netty的Android局域网IP电话

The Original Free and Open Source Game Control Panel