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<h1 id="awesome-real-time-communications-awesome">Awesome Real Time
Communications <a href="https://awesome.re"><img
src="https://awesome.re/badge.svg" alt="Awesome" /></a></h1>
<blockquote>
<p>Protocols and methodology for near simultaneous exchange of media and
data.</p>
</blockquote>
<h2 id="contents">Contents</h2>
<ul>
<li><a href="#server-software">Server Software</a>
<ul>
<li><a href="#general-purpose">General Purpose</a></li>
<li><a href="#sip-servers">SIP Servers</a></li>
<li><a href="#media-servers">Media Servers</a></li>
<li><a href="#stunturn">STUN/TURN</a></li>
</ul></li>
<li><a href="#operations">Operations</a>
<ul>
<li><a href="#monitoring">Monitoring</a></li>
<li><a href="#testing">Testing</a></li>
<li><a href="#deployment">Deployment</a></li>
<li><a href="#webapi-interfaces">Web/API Interfaces</a></li>
<li><a href="#billing">Billing</a></li>
</ul></li>
<li><a href="#developer-resources">Developer Resources</a>
<ul>
<li><a href="#tutorials">Tutorials</a></li>
<li><a href="#javascript-libraries">JavaScript Libraries</a></li>
<li><a href="#cc-libraries">C/C++ Libraries</a></li>
<li><a href="#go-libraries">Go Libraries</a></li>
<li><a href="#php-libraries">PHP Libraries</a></li>
<li><a href="#python-libraries">Python Libraries</a></li>
<li><a href="#erlang-libraries">Erlang Libraries</a></li>
<li><a href="#rust-libraries">Rust Libraries</a></li>
<li><a href="#dart-libraries">Dart Libraries</a></li>
</ul></li>
<li><a href="#blogs">Blogs</a></li>
<li><a href="#discussion">Discussion</a></li>
<li><a href="#events">Events</a></li>
<li><a href="#related-lists">Related Lists</a></li>
<li><a href="#contribute">Contribute</a></li>
</ul>
<h2 id="server-software">Server Software</h2>
<h3 id="general-purpose">General Purpose</h3>
<ul>
<li><a href="http://freeswitch.org">FreeSWITCH</a> - Open source
multi-protocol, cross-platform and software switch.</li>
<li><a href="http://asterisk.org">Asterisk</a> - PBX framework
supporting multiple protocols and platforms.</li>
</ul>
<h3 id="sip-servers">SIP Servers</h3>
<ul>
<li><a href="http://www.kamailio.org">Kamailio</a> - Open source SIP
server widely deployed by carriers and providers. Formerly known as
OpenSER.</li>
<li><a href="http://www.opensips.org">OpenSIPS</a> - Open source SIP
server, tracing its roots in OpenSER (presently Kamailio).</li>
<li><a href="https://routr.io">Routr</a> - Lightweight SIP proxy,
location server, and registrar written in Node.js.</li>
<li><a href="https://github.com/sippy/b2bua">Sippy B2BUA</a> -
Back-to-back user agent server written in Python.</li>
<li><a
href="https://github.com/BelledonneCommunications/flexisip">Flexisip</a>
- SIP server suite comprising proxy, presence and group chat
functions.</li>
</ul>
<h3 id="media-servers">Media Servers</h3>
<ul>
<li><a href="https://janus.conf.meetecho.com">Janus</a> - Lightweight
open source, general purpose, WebRTC gateway.</li>
<li><a href="https://www.rtpproxy.org">RTPProxy</a> - General purpose
high performance RTP proxy.</li>
<li><a href="https://github.com/sipwise/rtpengine">RTP:Engine</a> - RTP
and UDP based media traffic proxy, usable as a kernel module.</li>
<li><a href="https://mediasoup.org">mediasoup</a> - Specialized WebRTC
conferencing system.</li>
<li><a href="https://github.com/sems-server/sems">SEMS</a> - Open source
media and application server for SIP based VoIP services.</li>
<li><a href="https://jitsi.org/projects">Jitsi</a> - A collection of RTC
open source projects, with a focus on conferencing software.</li>
</ul>
<h3 id="stunturn">STUN/TURN</h3>
<ul>
<li><a href="https://github.com/coturn/coturn">coturn</a> - Fully
featured TURN/STUN server supporting multiple platforms.</li>
<li><a href="https://github.com/jselbie/stunserver">STUNTMAN</a> - RFC
compliant open source STUN implementation.</li>
</ul>
<h2 id="operations">Operations</h2>
<h3 id="monitoring">Monitoring</h3>
<ul>
<li><a href="https://github.com/irontec/sngrep">sngrep</a> - Terminal
based SIP flow viewer.</li>
<li><a href="https://github.com/sipcapture/sipgrep">sipgrep</a> -
Console tool for sniffing, capturing and exploring SIP traffic.</li>
<li><a href="https://github.com/Naishy/rtpsplit">rtpbreak</a> - Detect,
reconstruct and analyze RTP sessions.</li>
<li><a href="https://github.com/sipcapture/homer">HOMER</a> -
Multi-protocol capturing and monitoring framework for RTC.</li>
<li><a href="https://github.com/webrtc/testrtc">WebRTC
Troubleshooter</a> - Self-hosted one stop client-side WebRTC
troubleshooter.</li>
<li><a
href="https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice">Trickle
ICE</a> - Exposes client-side NAT traversal debug data.</li>
<li><a href="https://sip3.io">SIP3</a> - VoIP &amp; RTC traffic
monitoring and analysis platform.</li>
</ul>
<h3 id="testing">Testing</h3>
<ul>
<li><a href="http://sipp.sourceforge.net">SIPp</a> - Traffic generator
for the SIP protocol.</li>
<li><a
href="https://github.com/EnableSecurity/sipvicious">SIPVicious</a> -
Suite of security tools that can be used to audit SIP based VoIP
systems.</li>
<li><a href="https://github.com/nils-ohlmeier/sipsak">sipsak</a> - SIP
stress and diagnostics utility.</li>
<li><a href="https://github.com/miconda/sipexer">sipexer</a> - Modern
and flexible SIP command line tool.</li>
</ul>
<h3 id="deployment">Deployment</h3>
<ul>
<li><a href="https://github.com/rtckit/slimswitch">slimswitch</a> -
Tooling for creating lean secure FreeSWITCH Docker images.</li>
</ul>
<h3 id="webapi-interfaces">Web/API Interfaces</h3>
<ul>
<li><a href="https://eqivo.org">Eqivo</a> - Open source
programmable-voice/telephony API platform.</li>
<li><a href="https://www.2600hz.org">Kazoo</a> - Carrier-grade VoIP API
platform using FreeSWITCH and Kamailio.</li>
<li><a href="https://www.fusionpbx.com">FusionPBX</a> - Multitenant
system built on top of FreeSWITCH.</li>
<li><a href="https://www.freepbx.org">FreePBX</a> - Web Manager for
Asterisk.</li>
<li><a href="https://github.com/fonoster/fonoster">Fonoster</a> -
Telecommunication stack built with Node.js.</li>
<li><a href="https://wazo-platform.org">Wazo</a> - VoIP API platform
built on top of Asterisk, Kamailio and RTPEngine.</li>
<li><a href="https://www.jambonz.org">jambonz</a> - Open source CPaaS
built for communications service providers.</li>
<li><a href="https://github.com/irontec/ivozprovider">IVOZ Provider</a>
- Multitenant solution for VoIP telephony providers.</li>
</ul>
<h3 id="billing">Billing</h3>
<ul>
<li><a href="http://cgrates.org">CGRateS</a> - Carrier grade open source
billing/rating server.</li>
<li><a href="http://www.asterisk2billing.org">A2Billing</a> - Billing
system for Asterisk for multiple applications.</li>
<li><a
href="https://github.com/mwolff44/pyfreebilling">PyFreeBilling</a> -
Wholesale billing platform for Kamailio and FreeSWITCH.</li>
</ul>
<h2 id="developer-resources">Developer Resources</h2>
<h3 id="tutorials">Tutorials</h3>
<ul>
<li><a href="https://webrtc.org">Official Website</a> - Entry level
WebRTC resources.</li>
<li><a
href="https://www.html5rocks.com/en/tutorials/webrtc/basics">Getting
Started With WebRTC</a> - WebRTC tutorial by HTML5 Rocks.</li>
<li><a href="https://webrtc.github.io/samples">WebRTC Samples</a> -
Collection of samples demonstrating various parts of the WebRTC
APIs.</li>
<li><a href="https://www.webrtc-experiment.com">WebRTC Experiments</a> -
Comprehensive list of samples by Muaz Khan.</li>
<li><a
href="https://codelabs.developers.google.com/codelabs/webrtc-web">Interactive
Codelab</a> - 30 minutes step by step live tutorial by Google.</li>
</ul>
<h3 id="javascript-libraries">JavaScript Libraries</h3>
<ul>
<li><a href="https://drachtio.org">drachtio</a> - Node.js SIP server
framework.</li>
<li><a href="https://github.com/webrtcHacks/adapter">adapter.js</a> -
JavaScript shim for abstracting WebRTC spec changes and
inconsistencies.</li>
<li><a href="http://jssip.net">JsSIP</a> - Lightweight open source
JavaScript SIP library.</li>
<li><a href="https://www.doubango.org/sipml5">sipML5</a> - Open source
JavaScript SIP client with WebRTC media stack.</li>
<li><a href="https://github.com/feross/simple-peer">simple-peer</a> -
WebRTC video, voice, and data channels abstraction for Node.js and the
browser.</li>
<li><a href="https://github.com/coast-team/netflux">Netflux</a> -
Isomorphic JavaScript peer to peer transport API for client and
server.</li>
<li><a href="https://peerjs.com">PeerJS</a> - Data and media
peer-to-peer connection API implemented over WebRTC.</li>
</ul>
<h3 id="cc-libraries">C/C++ Libraries</h3>
<ul>
<li><a href="https://github.com/creytiv/re">libre</a> - Portable SIP
Stack along with companion libraries for media handling, STUN/TURN and a
modular user agent.</li>
<li><a href="https://www.pjsip.org">PJSIP</a> - Multi-protocol RTC
library written in C.</li>
<li><a href="http://savannah.nongnu.org/projects/exosip">eXosip</a> -
eXtended osip is a mature C library for abstracting the SIP
protocol.</li>
<li><a
href="https://github.com/paullouisageneau/libdatachannel">libdatachannel</a>
- Standalone WebRTC DataChannels C++ implementation.</li>
<li><a href="https://github.com/cisco/libsrtp">libSRTP</a> - Secure
Real-time Transport Protocol (SRTP) library for C.</li>
<li><a href="https://github.com/sctplab/usrsctp">usrsctp</a> - Portable
Stream Control Transmission Protocol (SCTP) user-land stack.</li>
<li><a href="https://github.com/rawrtc/rawrtc">rawrtc</a> - WebRTC and
ORTC library with a small footprint.</li>
<li><a href="https://github.com/joegen/oss_core">OSS Core</a> - General
purpose C++ library for Real Time Communications.</li>
<li><a href="https://01.org/open-webrtc-toolkit">Open WebRTC Toolkit</a>
- WebRTC development toolkit with bindings for multiple platforms.</li>
<li><a href="https://github.com/freeswitch/sofia-sip">Sofia-SIP</a> -
Open source SIP library used by FreeSWITCH.</li>
</ul>
<h3 id="go-libraries">Go Libraries</h3>
<ul>
<li><a href="https://pion.ly">Pion</a> - Extensive software stack for
WebRTC written in Go.</li>
<li><a href="https://github.com/StefanKopieczek/gossip">gossip</a> - SIP
stack for stateful user agents written in Go.</li>
<li><a href="https://github.com/marv2097/siprocket">siprocket</a> - Fast
SIP and SDP packet parser.</li>
<li><a href="https://github.com/fiorix/go-diameter">go-diameter</a> -
RFC compliant Diameter protocol library.</li>
</ul>
<h3 id="php-libraries">PHP Libraries</h3>
<ul>
<li><a href="https://github.com/rtckit/php-sip">RTCKit/SIP</a> - RFC
3261 compliant SIP parsing and rendering library for PHP 7.4+.</li>
</ul>
<h3 id="python-libraries">Python Libraries</h3>
<ul>
<li><a href="https://github.com/aiortc/aiortc">aiortc</a> - WebRTC and
ORTC implementation for Python using asyncio.</li>
<li><a href="https://github.com/hyperioxx/Katari">Katari</a> - SIP stack
application framework.</li>
<li><a
href="https://github.com/ambianic/peerjs-python">peerjs-python</a> -
Python port of the PeerJS peer-to-peer connection library.</li>
</ul>
<h3 id="erlang-libraries">Erlang Libraries</h3>
<ul>
<li><a href="https://github.com/NetComposer/nksip">NkSIP</a> -
Extendable SIP server framework.</li>
<li><a href="https://github.com/poroh/ersip">ersip</a> - Library
comprising building blocks for SIP applications.</li>
</ul>
<h3 id="rust-libraries">Rust Libraries</h3>
<ul>
<li><a href="https://docs.rs/libsip/0.2.4/libsip">libsip</a> - SIP
implementation, with a focus towards softphone clients.</li>
<li><a href="https://github.com/armatusmiles/sipcore">sipcore</a> - Rust
framework for creating SIP applications.</li>
<li><a href="https://github.com/rtcrs/webrtc">rtcrs/webrtc</a> - WebRTC
stack, supporting SDP, RTP, RTCP and SRTP.</li>
</ul>
<h3 id="dart-libraries">Dart Libraries</h3>
<ul>
<li><a href="https://github.com/cloudwebrtc/dart-sip-ua">dart-sip-ua</a>
- Dart-lang port of JsSIP, capable of SIP over WebSocket.</li>
</ul>
<h2 id="blogs">Blogs</h2>
<ul>
<li><a href="https://bloggeek.me/blog">BlogGeekMe</a> - Blog by Tsahi
Levent-Levi with a strong focus on WebRTC.</li>
<li><a href="https://andrewjprokop.wordpress.com">SIP Adventures</a> -
Unified communications blog by Andrew Prokop.</li>
<li><a href="https://webrtchacks.com">WebRTCHacks</a> - WebRTC blog by
independent technologists.</li>
</ul>
<h2 id="discussion">Discussion</h2>
<ul>
<li><a href="https://signalwire.community">FreeSWITCH Slack</a> - Join
#freeswitch and #freeswitch-dev for user and developer support.</li>
<li><a
href="https://groups.google.com/forum/?fromgroups#!forum/discuss-webrtc">discuss-webrtc</a>
- Developer oriented Google Group for WebRTC discussions.</li>
</ul>
<h2 id="events">Events</h2>
<ul>
<li><a href="http://cluecon.com">ClueCon</a> - Annual conference held in
Chicago for telecommunications developers. Birthplace of
FreeSWITCH.</li>
<li><a href="https://www.kamailioworld.com">Kamailio World</a> - Berlin
hosted annual event focused on Kamailio as well as VoIP, WebRTC, IMS,
VoLTE and more.</li>
<li><a
href="https://www.asterisk.org/community/astricon-user-conference">AstriCon</a>
- Asterisk focus event held every year across the US.</li>
<li><a href="https://commcon.xyz">CommCon</a> - Annual conference held
in the UK focused on telecommunications in general and WebRTC in
particular.</li>
<li><a href="https://www.opensips.org/events">OpenSIPS Summit</a> -
Meeting place for the OpenSIPS community.</li>
<li><a href="https://krankygeek.com">Kranky Geek</a> - AI and RTC event
in San Francisco.</li>
<li><a href="https://fosdem.org">FOSDEM</a> - Free event for software
developers, with a RTC component, held every year in Europe.</li>
<li><a href="https://www.januscon.it">JanusCon</a> - JanusCon is a live
event for Janus and RTC implementers.</li>
<li><a href="https://tadhack.com">TADHack</a> - Global hackathon focused
on programmable communications.</li>
</ul>
<h2 id="related-lists">Related Lists</h2>
<ul>
<li><a href="https://github.com/rtckit/awesome-ript">Awesome RIPT</a> -
Real Time Internet Peering for Telephony.</li>
<li><a
href="https://github.com/EnableSecurity/awesome-rtc-hacking">Awesome RTC
Hacking</a> - Real Time Communications hacking and penetration testing
resources.</li>
<li><a href="https://github.com/calee0219/awesome-5g">Awesome 5G</a> -
5G frameworks, libraries, software and resources.</li>
<li><a href="https://github.com/W00t3k/Awesome-Cellular-Hacking">Awesome
Cellular Hacking</a> - Research resources in the 3G/4G/5G Cellular
security space.</li>
<li><a href="https://github.com/ravens/awesome-telco">Awesome Telco</a>
- Telco resources and projects.</li>
<li><a href="https://github.com/miconda/sip-resources">SIP Resources</a>
- Useful SIP resources curated by Kamailio’s head developer.</li>
</ul>
<h2 id="contribute">Contribute</h2>
<p>Contributions welcome! Read the <a
href="CONTRIBUTING.md">contribution guidelines</a> first.</p>
<p><a href="https://github.com/rtckit/awesome-rtc">rtc.md Github</a></p>