Awesome Real Time Communications

Protocols and methodology for near simultaneous exchange of media and data.

Contents

• Server Software
• General Purpose
• SIP Servers
• Media Servers
• STUN/TURN
• Operations
• Monitoring
• Testing
• Deployment
• Web/API Interfaces
• Billing
• Developer Resources
• Tutorials
• JavaScript Libraries
• C/C++ Libraries
• Go Libraries
• PHP Libraries
• Python Libraries
• Erlang Libraries
• Rust Libraries
• Dart Libraries
• Blogs
• Discussion
• Events
• Related Lists
• Contribute

Server Software

General Purpose

• FreeSWITCH - Open source multi-protocol, cross-platform and software switch.
• Asterisk - PBX framework supporting multiple protocols and platforms.

SIP Servers

• Kamailio - Open source SIP server widely deployed by carriers and providers. Formerly known as OpenSER.
• OpenSIPS - Open source SIP server, tracing its roots in OpenSER (presently Kamailio).
• Routr - Lightweight SIP proxy, location server, and registrar written in Node.js.
• Sippy B2BUA - Back-to-back user agent server written in Python.
• Flexisip - SIP server suite comprising proxy, presence and group chat functions.

Media Servers

• Janus - Lightweight open source, general purpose, WebRTC gateway.
• RTPProxy - General purpose high performance RTP proxy.
• RTP:Engine - RTP and UDP based media traffic proxy, usable as a kernel module.
• mediasoup - Specialized WebRTC conferencing system.
• SEMS - Open source media and application server for SIP based VoIP services.
• Jitsi - A collection of RTC open source projects, with a focus on conferencing software.

STUN/TURN

• coturn - Fully featured TURN/STUN server supporting multiple platforms.
• eturnal - Modern and scalable STUN/TURN server written in Erlang.
• STUNTMAN - RFC compliant open source STUN implementation.

Operations

Monitoring

• sngrep - Terminal based SIP flow viewer.
• sipgrep - Console tool for sniffing, capturing and exploring SIP traffic.
• rtpbreak - Detect, reconstruct and analyze RTP sessions.
• HOMER - Multi-protocol capturing and monitoring framework for RTC.
• WebRTC Troubleshooter - Self-hosted one stop client-side WebRTC troubleshooter.
• Trickle ICE - Exposes client-side NAT traversal debug data.
• SIP3 - VoIP & RTC traffic monitoring and analysis platform.

Testing

• SIPp - Traffic generator for the SIP protocol.
• SIPVicious - Suite of security tools that can be used to audit SIP based VoIP systems.
• sipsak - SIP stress and diagnostics utility.
• sipexer - Modern and flexible SIP command line tool.

Deployment

• slimswitch - Tooling for creating lean secure FreeSWITCH Docker images.

Web/API Interfaces

• Eqivo - Open source programmable-voice/telephony API platform.
• Kazoo - Carrier-grade VoIP API platform using FreeSWITCH and Kamailio.
• FusionPBX - Multitenant system built on top of FreeSWITCH.
• FreePBX - Web Manager for Asterisk.
• Fonoster - Telecommunication stack built with Node.js.
• Wazo - VoIP API platform built on top of Asterisk, Kamailio and RTPEngine.
• jambonz - Open source CPaaS built for communications service providers.
• IVOZ Provider - Multitenant solution for VoIP telephony providers.
• Sayna - Real-time speech infrastructure for voice AI with WebSocket streaming, SIP telephony and pluggable STT/TTS providers.