voip

Pure Go implementation of the WebRTC API

即时通讯(IM)系统

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.

Async PHP client API for the telegram MTProto protocol

The official Asterisk Project repository.

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✨

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)

bulk outbound calls with automatic Whisper transcription via Telnyx

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

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.

I'm sorry, Dave. I'm afraid I can't let that spam call through. — Local AI answering service with voice cloning, call recording, and sub-second latency. Runs entirely on your GPU. No cloud. No per-minute billing. Just HAL.