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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.

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General Purpose

Open source multi-protocol, cross-platform and software switch.

PBX framework supporting multiple protocols and platforms.

SIP Servers

Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) Back-to-back user agent (B2BUA) server software.

106
53
92d
BSD-2-Clause

Linphone.org mirror for flexisip (git://git.linphone.org/flexisip.git)

75
45
14d
AGPL-3.0

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).

Lightweight SIP proxy, location server, and registrar written in Node.js.

Media Servers

The Sipwise media proxy for Kamailio

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245
14d
GPL-3.0

Sip Express Media Server

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63
18d
n/a

Lightweight open source, general purpose, WebRTC gateway.

General purpose high performance RTP proxy.

Specialized WebRTC conferencing system.

STUN/TURN

coturn TURN server project

5.99K
1.28K
45d
n/a

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.

981
277
10m
Apache-2.0

Monitoring

Ncurses SIP Messages flow viewer

603
125
27d
n/a

SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console

131
39
80d
GPL-3.0

RTP stream extractor

5
6
4y 31d
GPL-2.0

HOMER - 100% Open-Source SIP / VoIP Packet Capture & Monitoring

842
176
67d
AGPL-3.0

One stop client-side WebRTC troubleshooter.

Exposes client-side NAT traversal debug data.

VoIP & RTC traffic monitoring and analysis platform.

Testing

SIPVicious OSS is a set of security tools that can be used to audit SIP based VoIP systems.

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128
24d
n/a

SIP swiss army knife

75
27
55d
GPL-2.0

Traffic generator for the SIP protocol.

Web/API Interfaces

Carrier-grade VoIP API platform using FreeSWITCH and Kamailio.

Multitenant system built on top of FreeSWITCH.

Web Manager for Asterisk.

Billing

Routing and rating VoIP application for service providers - API based - AGPL v3 - Based on kamailio

71
41
9m
AGPL-3.0

Carrier grade open source billing/rating server.

Billing system for Asterisk for multiple applications.

Tutorials

Entry level WebRTC resources.

Collection of samples demonstrating various parts of the WebRTC APIs.

Comprehensive list of samples by Muaz Khan.

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:

2.69K
718
18d
BSD-3-Clause

๐Ÿ“ก Simple WebRTC video, voice, and data channels

5.12K
772
79d
MIT

JavaScript client and server side transport API based on WebRTC & WebSocket

188
12
2y 5m
AGPL-3.0

Node.js SIP server framework.

Lightweight open source JavaScript SIP library.

Open source JavaScript SIP client with WebRTC media stack.

Data and media peer-to-peer connection API implemented over WebRTC.

C/C++ Libraries

Generic library for real-time communications with async IO support

435
176
5m
n/a

C/C++ WebRTC Data Channels and Media Transport standalone library

315
56
15d
LGPL-2.1

Library for SRTP (Secure Realtime Transport Protocol)

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340
23d
n/a

A portable SCTP userland stack

397
202
30d
BSD-3-Clause

WebRTC and ORTC with a little bit of RAWR!

317
27
1y 6m
BSD-2-Clause

SIP Session Border Controller Library

19
7
1y 9m
n/a

Multi-protocol RTC library written in C.

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

277
95
4y 4m
LGPL-2.1

Fast SIP and SDP Parser

57
17
1y 6m
MIT

Diameter stack and Base Protocol (RFC 6733) for the Go programming language

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88
112d
n/a

Extensive software stack for WebRTC written in Go.

PHP Libraries

SIP Parsing/Rendering Library for PHP

3
0
4m
MIT

Python Libraries

WebRTC and ORTC implementation for Python using asyncio

2.22K
363
30d
BSD-3-Clause

Katari - Python Session Initiated Protocol Framework

16
4
1y 99d
BSD-3-Clause

Python port of PeerJS client

39
12
19d
Apache-2.0

Erlang Libraries

Erlang SIP application server

321
114
2y 48d
Apache-2.0

Erlang SIP

88
15
36d
MIT

Rust Libraries

Rust framework for creating SIP applications

13
0
94d
MIT

A pure Rust implementation of WebRTC API

716
26
21d
MIT

SIP implementation, with a focus towards softphone clients.

Dart Libraries

A dart-lang version of the SIP UA stack.

131
57
24d
MIT

Blogs

Blog by Tsahi Levent-Levi with a strong focus on WebRTC.

Unified communications blog by Andrew Prokop.

WebRTC blog by independent technologists.

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.

Meeting place for the OpenSIPS community.

AI and RTC event in San Francisco.

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

8
0
4m
CC0-1.0

a list of awesome resources related to security and hacking of VoIP, WebRTC and VoLTE

94
11
4m
CC0-1.0

Awesome lists about 5G projects.

40
10
4m
CC0-1.0

Awesome-Cellular-Hacking

1.42K
323
5m
n/a

A curated list of telco resources and projects

165
37
6m
MIT