Awesome Real Time Communications – Massive Collection of Resources

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
  • 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

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.
• 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 – 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.

Web/API Interfaces

• Kazoo – Carrier-grade VoIP API platform using FreeSWITCH and Kamailio.
• FusionPBX – Multitenant system built on top of FreeSWITCH.
• FreePBX – Web Manager for Asterisk.

Billing

• CGRateS – Carrier grade open source billing/rating server.
• A2Billing – Billing system for Asterisk for multiple applications.
• PyFreeBilling – Wholesale billing platform for Kamailio and FreeSWITCH.

Developer Resources

Tutorials

• 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

• 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.

C/C++ Libraries

• 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.

Go Libraries

• 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

• RTCKit/SIP – RFC 3261 compliant SIP parsing and rendering library for PHP 7.4+.

Python Libraries

• 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

• NkSIP – Extendable SIP server framework.
• ersip – Library comprising building blocks for SIP applications.

Rust Libraries

• 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

• dart-sip-ua – Dart-lang port of JsSIP, capable of SIP over WebSocket.

Blogs

• 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

• FreeSWITCH Slack – Join #freeswitch and #freeswitch-dev for user and developer support.
• discuss-webrtc – Developer oriented Google Group for WebRTC discussions.

Events

• 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.

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