voip

Pure Go implementation of the WebRTC API

即时通讯(IM)系统

Mumble is an open-source, low-latency, high quality voice chat software.

🚀 The open-source alternative to Twilio.

Robust, Ubiquitous and Massively Scalable Messaging Platform (XMPP, MQTT, SIP Server)

qTox is a chat, voice, video, and file transfer IM client using the encrypted peer-to-peer Tox protocol.

A pure Rust implementation of WebRTC

WebRTC plugin for Flutter Mobile/Desktop/Web

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.

🚀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：

Async PHP client API for the telegram MTProto protocol

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✨

The official Asterisk Project repository.

🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 8k, compatible with all browsers and platforms.

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

Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -

PJSIP project

A simple, intuitive, and powerful JavaScript signaling library

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

HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring