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.

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

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

SIP support for AsyncIO (DEPRECATED)

VOIP Security Audit Framework

100% Open-Source Packet Capture Agent for HEP

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

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

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

A dart-lang version of the SIP UA stack.

A simple, intuitive, and powerful JavaScript signaling library

A fully featured browser based WebRTC SIP phone for Asterisk

Set of tools to audit SIP based VoIP Systems

🤙Learning VoIP, RTP, pjsip and SIP

A Go implementation of STUN

Python SIP Packet forging tool

HEP Capture Server

WebRTC plugin for Flutter Mobile/Desktop/Web

Telegram <-> SIP voice gateway

IVOZ Provider - Multitenant solution for VoIP telephony providers