Real-Time Communications
Contents_Index
General Purpose
2_ENTRIES- FreeSWITCH
Open source multi-protocol, cross-platform and software switch.
- Asterisk
PBX framework supporting multiple protocols and platforms.
SIP Servers
5_ENTRIES- 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
6_ENTRIES- 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
3_ENTRIESMonitoring
7_ENTRIES- 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
4_ENTRIES- 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
1_ENTRIES- slimswitch
Tooling for creating lean secure FreeSWITCH Docker images.
Web/API Interfaces
9_ENTRIES- 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.
Billing
3_ENTRIES- CGRateS
Carrier grade open source billing/rating server.
- A2Billing
Billing system for Asterisk for multiple applications.
- PyFreeBilling
Wholesale billing platform for Kamailio and FreeSWITCH.
Tutorials
5_ENTRIES- Official Website
Entry level WebRTC resources.
- Getting Started With WebRTC
WebRTC tutorial by HTML5 Rocks.
- WebRTC Samples
Collection of samples demonstrating various parts of the WebRTC APIs.
- WebRTC Experiments
Comprehensive list of samples by Muaz Khan.
- Interactive Codelab
30 minutes step by step live tutorial by Google.
JavaScript Libraries
8_ENTRIES- drachtio
Node.js SIP server framework.
- adapter.js
JavaScript shim for abstracting WebRTC spec changes and inconsistencies.
- JsSIP
Lightweight open source JavaScript SIP library.
- sipML5
Open source JavaScript SIP client with WebRTC media stack.
- simple-peer
WebRTC video, voice, and data channels abstraction for Node.js and the browser.
- Netflux
Isomorphic JavaScript peer to peer transport API for client and server.
- PeerJS
Data and media peer-to-peer connection API implemented over WebRTC.
- Socio
A WebSocket Real-Time Communication (RTC) API framework. Realtime Front-end, Back-end reactivity.
C/C++ Libraries
10_ENTRIES- libre
Portable SIP Stack along with companion libraries for media handling, STUN/TURN and a modular user agent.
- PJSIP
Multi-protocol RTC library written in C.
- eXosip
eXtended osip is a mature C library for abstracting the SIP protocol.
- libdatachannel
Standalone WebRTC DataChannels C++ implementation.
- libSRTP
Secure Real-time Transport Protocol (SRTP) library for C.
- usrsctp
Portable Stream Control Transmission Protocol (SCTP) user-land stack.
- rawrtc
WebRTC and ORTC library with a small footprint.
- OSS Core
General purpose C++ library for Real Time Communications.
- Open WebRTC Toolkit
WebRTC development toolkit with bindings for multiple platforms.
- Sofia-SIP
Open source SIP library used by FreeSWITCH.
Go Libraries
4_ENTRIES- Pion
Extensive software stack for WebRTC written in Go.
- gossip
SIP stack for stateful user agents written in Go.
- siprocket
Fast SIP and SDP packet parser.
- go-diameter
RFC compliant Diameter protocol library.
PHP Libraries
1_ENTRIES- RTCKit/SIP
RFC 3261 compliant SIP parsing and rendering library for PHP 7.4+.
Python Libraries
3_ENTRIES- aiortc
WebRTC and ORTC implementation for Python using asyncio.
- Katari
SIP stack application framework.
- peerjs-python
Python port of the PeerJS peer-to-peer connection library.
Erlang Libraries
2_ENTRIESRust Libraries
3_ENTRIES- libsip
SIP implementation, with a focus towards softphone clients.
- sipcore
Rust framework for creating SIP applications.
- rtcrs/webrtc
WebRTC stack, supporting SDP, RTP, RTCP and SRTP.
Dart Libraries
1_ENTRIES- dart-sip-ua
Dart-lang port of JsSIP, capable of SIP over WebSocket.
Blogs
3_ENTRIES- BlogGeekMe
Blog by Tsahi Levent-Levi with a strong focus on WebRTC.
- SIP Adventures
Unified communications blog by Andrew Prokop.
- WebRTCHacks
WebRTC blog by independent technologists.
Discussion
2_ENTRIES- FreeSWITCH Slack
Join #freeswitch and #freeswitch-dev for user and developer support.
- discuss-webrtc
Developer oriented Google Group for WebRTC discussions.
Events
9_ENTRIES- ClueCon
Annual conference held in Chicago for telecommunications developers. Birthplace of FreeSWITCH.
- Kamailio World
Berlin hosted annual event focused on Kamailio as well as VoIP, WebRTC, IMS, VoLTE and more.
- AstriCon
Asterisk focus event held every year across the US.
- CommCon
Annual conference held in the UK focused on telecommunications in general and WebRTC in particular.
- OpenSIPS Summit
Meeting place for the OpenSIPS community.
- Kranky Geek
AI and RTC event in San Francisco.
- FOSDEM
Free event for software developers, with a RTC component, held every year in Europe.
- JanusCon
JanusCon is a live event for Janus and RTC implementers.
- TADHack
Global hackathon focused on programmable communications.