awesome-rtc
:satellite: A curated list of awesome Real Time Communications resources
https://github.com/rtckit/awesome-rtc
Last synced: 9 days ago
JSON representation
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Developer Resources
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C/C++ Libraries
- eXosip - eXtended osip is a mature C library for abstracting the SIP protocol.
- libre - Portable SIP Stack along with companion libraries for media handling, STUN/TURN and a modular user agent.
- 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.
- Sofia-SIP - Open source SIP library used by FreeSWITCH.
- icey - C++20 WebRTC media runtime with FFmpeg pipeline, Symple signalling, and RFC 5766 TURN.
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Erlang Libraries
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Go Libraries
- gossip - SIP stack for stateful user agents written in Go.
- siprocket - Fast SIP and SDP packet parser.
- go-diameter - RFC compliant Diameter protocol library.
- Pion - Extensive software stack for WebRTC written in Go.
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JavaScript Libraries
- adapter.js - JavaScript shim for abstracting WebRTC spec changes and inconsistencies.
- 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.
- Socio - A WebSocket Real-Time Communication (RTC) API framework. Realtime Front-end, Back-end reactivity.
- drachtio - Node.js SIP server framework.
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PHP Libraries
- RTCKit/SIP - RFC 3261 compliant SIP parsing and rendering library for PHP 7.4+.
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Python Libraries
- Katari - SIP stack application framework.
- aiortc - WebRTC and ORTC implementation for Python using asyncio.
- peerjs-python - Python port of the PeerJS peer-to-peer connection library.
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Tutorials
- Interactive Codelab - 30 minutes step by step live tutorial by Google.
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Discussion
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Dart Libraries
- discuss-webrtc - Developer oriented Google Group for WebRTC discussions.
- FreeSWITCH Slack - Join #freeswitch and #freeswitch-dev for user and developer support.
- FreeSWITCH Slack - Join #freeswitch and #freeswitch-dev for user and developer support.
- FreeSWITCH Slack - Join #freeswitch and #freeswitch-dev for user and developer support.
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Events
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Dart Libraries
- TADHack - Global hackathon focused on programmable communications.
- Kranky Geek - AI and RTC event in San Francisco.
- CommCon - Annual conference held in the UK focused on telecommunications in general and WebRTC in particular.
- 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.
- JanusCon - JanusCon is a live event for Janus and RTC implementers.
- TADHack - Global hackathon focused on programmable communications.
- Kamailio World - Berlin hosted annual event focused on Kamailio as well as VoIP, WebRTC, IMS, VoLTE and more.
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Operations
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Billing
- CGRateS - Carrier grade open source billing/rating server.
- PyFreeBilling - Wholesale billing platform for Kamailio and FreeSWITCH.
- CGRateS - Carrier grade open source billing/rating server.
- A2Billing - Billing system for Asterisk for multiple applications.
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Deployment
- slimswitch - Tooling for creating lean secure FreeSWITCH Docker images.
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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.
- SIP3 - VoIP & RTC traffic monitoring and analysis platform.
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Testing
- 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.
- SIPp - Traffic generator for the SIP protocol.
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Web/API Interfaces
- Kazoo - Carrier-grade VoIP API platform using FreeSWITCH and Kamailio.
- Fonoster - Telecommunication stack built with Node.js.
- IVOZ Provider - Multitenant solution for VoIP telephony providers.
- FreePBX - Web Manager for Asterisk.
- Sayna - Real-time speech infrastructure for voice AI with WebSocket streaming, SIP telephony and pluggable STT/TTS providers.
- Kazoo - Carrier-grade VoIP API platform using FreeSWITCH and Kamailio.
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Related Lists
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Dart Libraries
- Awesome RIPT - Real Time Internet Peering for Telephony.
- Awesome RTC Hacking - Real Time Communications hacking and penetration testing resources.
- Awesome 5G - 5G frameworks, libraries, software and resources.
- Awesome Cellular Hacking - Research resources in the 3G/4G/5G Cellular security space.
- Awesome Telco - Telco resources and projects.
- SIP Resources - Useful SIP resources curated by Kamailio's head developer.
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Server Software
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General Purpose
- Asterisk - PBX framework supporting multiple protocols and platforms.
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Media Servers
- RTP:Engine - RTP and UDP based media traffic proxy, usable as a kernel module.
- SEMS - Open source media and application server for SIP based VoIP services.
- RTPProxy - General purpose high performance RTP proxy.
- LiveKit - Open-source WebRTC infrastructure for building scalable real-time audio and video applications.
- LiveKit - Open-source WebRTC infrastructure for building scalable real-time audio and video applications.
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SIP Servers
- Sippy B2BUA - Back-to-back user agent server written in Python.
- Flexisip - SIP server suite comprising proxy, presence and group chat functions.
- Routr - Lightweight SIP proxy, location server, and registrar written in Node.js.
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STUN/TURN
- coturn - Fully featured TURN/STUN server supporting multiple platforms.
- STUNTMAN - RFC compliant open source STUN implementation.
- eturnal - Modern and scalable STUN/TURN server written in Erlang.
- natcheck - NAT type diagnosis CLI. Probes STUN servers, classifies mapping behaviour per RFC 5780, and reports a WebRTC direct-P2P forecast.
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Programming Languages
Categories
Sub Categories
Keywords
webrtc
14
sip
14
voip
11
telephony
6
p2p
4
security
3
python
3
rtp
3
peer-connection
2
c
2
ortc
2
data-channel
2
networking
2
turn
2
flow
2
hep
2
kamailio
2
peer-to-peer
2
pcap
2
hacking
2
awesome-lists
2
sdp
2
telecommunications
2
awesome
2
sctp
2
nodejs
2
server
2
go
2
javascript
2
websocket
2
cloud
1
webrtc-adapter
1
polyfill
1
data-channels
1
svcrack
1
security-tools
1
password-cracker
1
hacking-tools
1
audit-sip
1
scale
1
provider
1
platform
1
pbx
1
multitenant
1
ucaas
1
typescript
1
twilio
1
programmable-voice
1
kubernetes
1
customer-experience
1