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Awesome Real-Time Communications
A curated list of awesome Real Time Communications resources
Here you can see meta information about this topic like the time we last updated this page, the original creator of the awesome list and a link to the original GitHub repository.
Thank you rtckit & contributors
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General Purpose
SIP Servers
Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) Back-to-back user agent (B2BUA) server software.
Linphone.org mirror for flexisip (git://git.linphone.org/flexisip.git)
Open source SIP server widely deployed by carriers and providers. Formerly known as OpenSER.
Open source SIP server, tracing its roots in OpenSER (presently Kamailio).
Media Servers
The Sipwise media proxy for Kamailio
Sip Express Media Server
STUN/TURN
coturn TURN server project
Version 1.2. This is the source code to STUNTMAN - an open source STUN server and client code by john selbie. Compliant with the latest RFCs including 5389, 5769, and 5780. Also includes backwards compatibility for RFC 3489. Compiles on Linux, MacOS, BSD, Solaris, and Win32 with Cygwin. Windows binaries avaialble from www.stunprotocol.org.
Monitoring
Ncurses SIP Messages flow viewer
SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console
RTP stream extractor
HOMER - 100% Open-Source SIP / VoIP Packet Capture & Monitoring
Exposes client-side NAT traversal debug data.
Testing
SIPVicious OSS is a set of security tools that can be used to audit SIP based VoIP systems.
SIP swiss army knife
Web/API Interfaces
Billing
Routing and rating VoIP application for service providers - API based - AGPL v3 - Based on kamailio
Tutorials
WebRTC tutorial by HTML5 Rocks.
Collection of samples demonstrating various parts of the WebRTC APIs.
30 minutes step by step live tutorial by Google.
JavaScript Libraries
Shim to insulate apps from spec changes and prefix differences. Latest adapter.js release:
📡 Simple WebRTC video, voice, and data channels
JavaScript client and server side transport API based on WebRTC & WebSocket
C/C++ Libraries
Generic library for real-time communications with async IO support
C/C++ WebRTC Data Channels and Media Transport standalone library
Library for SRTP (Secure Realtime Transport Protocol)
A portable SCTP userland stack
WebRTC and ORTC with a little bit of RAWR!
SIP Session Border Controller Library
eXtended osip is a mature C library for abstracting the SIP protocol.
WebRTC development toolkit with bindings for multiple platforms.
Go Libraries
SIP stack in Golang
Fast SIP and SDP Parser
Diameter stack and Base Protocol (RFC 6733) for the Go programming language
PHP Libraries
SIP Parsing/Rendering Library for PHP
Python Libraries
WebRTC and ORTC implementation for Python using asyncio
Katari - Python Session Initiated Protocol Framework
Python port of PeerJS client
Erlang Libraries
Erlang SIP application server
Erlang SIP
Rust Libraries
Rust framework for creating SIP applications
A pure Rust implementation of WebRTC API. Rewrite Pion WebRTC stack (http://Pion.ly) in Rust!
SIP implementation, with a focus towards softphone clients.
Dart Libraries
A dart-lang version of the SIP UA stack.
Blogs
Discussion
Join #freeswitch and #freeswitch-dev for user and developer support.
Developer oriented Google Group for WebRTC discussions.
Events
Annual conference held in Chicago for telecommunications developers. Birthplace of FreeSWITCH.
Berlin hosted annual event focused on Kamailio as well as VoIP, WebRTC, IMS, VoLTE and more.
Asterisk focus event held every year across the US.
Annual conference held in the UK focused on telecommunications in general and WebRTC in particular.
Free event for software developers, with a RTC component, held every year in Europe.
Related Lists
Curated list of Realtime Internet Peering for Telephony (RIPT) resources
a list of awesome resources related to security and hacking of VoIP, WebRTC and VoLTE
Awesome lists about 5G projects.
Awesome-Cellular-Hacking
A curated list of telco resources and projects