Real-Time Communications

Back-to-back user agent server written in Python.

SIP server suite comprising proxy, presence and group chat functions.

RTP and UDP based media traffic proxy, usable as a kernel module.

Open source media and application server for SIP based VoIP services.

Fully featured TURN/STUN server supporting multiple platforms.

RFC compliant open source STUN implementation.

Terminal based SIP flow viewer.

Console tool for sniffing, capturing and exploring SIP traffic.

Detect, reconstruct and analyze RTP sessions.

Multi-protocol capturing and monitoring framework for RTC.

Self-hosted one stop client-side WebRTC troubleshooter.

Suite of security tools that can be used to audit SIP based VoIP systems.

SIP stress and diagnostics utility.

Modern and flexible SIP command line tool.

Tooling for creating lean secure FreeSWITCH Docker images.

Telecommunication stack built with Node.js.

Multitenant solution for VoIP telephony providers.

Real-time speech infrastructure for voice AI with WebSocket streaming, SIP telephony and pluggable STT/TTS providers.

Wholesale billing platform for Kamailio and FreeSWITCH.

JavaScript shim for abstracting WebRTC spec changes and inconsistencies.

WebRTC video, voice, and data channels abstraction for Node.js and the browser.

Isomorphic JavaScript peer to peer transport API for client and server.

A WebSocket Real-Time Communication (RTC) API framework. Realtime Front-end, Back-end reactivity.

Portable SIP Stack along with companion libraries for media handling, STUN/TURN and a modular user agent.

Standalone WebRTC DataChannels C++ implementation.

Secure Real-time Transport Protocol (SRTP) library for C.

Portable Stream Control Transmission Protocol (SCTP) user-land stack.

WebRTC and ORTC library with a small footprint.

General purpose C++ library for Real Time Communications.

Open source SIP library used by FreeSWITCH.

SIP stack for stateful user agents written in Go.

Fast SIP and SDP packet parser.

RFC compliant Diameter protocol library.

RFC 3261 compliant SIP parsing and rendering library for PHP 7.4+.

WebRTC and ORTC implementation for Python using asyncio.

SIP stack application framework.

Python port of the PeerJS peer-to-peer connection library.

Extendable SIP server framework.

Library comprising building blocks for SIP applications.

Rust framework for creating SIP applications.

WebRTC stack, supporting SDP, RTP, RTCP and SRTP.

Dart-lang port of JsSIP, capable of SIP over WebSocket.