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.