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Robust, Ubiquitous and Massively Scalable Messaging Platform (XMPP, MQTT, SIP Server)
Created
2010-02-08
9,419 commits to master branch, last one 3 days ago
Jitsi is an audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, IRC and many other useful features.
Created
2013-04-24
13,240 commits to master branch, last one about a month ago
WebRTC plugin for Flutter Mobile/Desktop/Web
Created
2018-03-07
1,102 commits to main branch, last one a day ago
FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a ...
Created
2018-10-24
35,766 commits to master branch, last one 12 days ago
RTSP/RTP/RTMP/FLV/HLS/MPEG-TS/MPEG-PS/MPEG-DASH/MP4/fMP4/MKV/WebM
Created
2014-01-03
1,425 commits to master branch, last one 7 days ago
Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
Created
2013-12-11
37,348 commits to master branch, last one a day ago
The official Asterisk Project repository.
Created
2015-04-11
33,884 commits to master branch, last one 14 days ago
🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 4k, compatible with all browsers and platforms.
Created
2021-08-17
1,499 commits to main branch, last one 21 hours ago
PJSIP project
Created
2016-01-24
6,469 commits to master branch, last one 2 days ago
A simple, intuitive, and powerful JavaScript signaling library
Created
2013-12-20
2,509 commits to main branch, last one about a year ago
Baresip is a modular SIP User-Agent with audio and video support
Created
2014-02-09
4,067 commits to main branch, last one 23 hours ago
HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
Created
2014-03-28
1,236 commits to homer10 branch, last one a day ago
⚡ The future of programmable SIP servers.
Created
2017-01-18
2,792 commits to main branch, last one about a month ago
A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
Created
2015-10-29
2,561 commits to master branch, last one about a month ago
Linphone.org mirror for linphone-android (https://gitlab.linphone.org/BC/public/linphone-android)
Created
2015-01-14
7,939 commits to master branch, last one 2 months ago
SIP softphone for Mac
Created
2011-01-15
2,451 commits to master branch, last one about a year ago
Pluggable WebRTC softphone and communication platform.
Created
2017-04-03
590 commits to develop branch, last one 5 years ago
Ncurses SIP Messages flow viewer
Created
2013-06-20
1,008 commits to master branch, last one about a month ago
SIPVicious OSS is a VoIP security testing toolset. It helps security teams, QA and developers test SIP-based VoIP systems and applications. This toolset is useful in simulating VoIP hacking attacks ag...
Created
2015-03-13
462 commits to master branch, last one about a year ago
Linux/Win/Docker/kubernetes/Chart/Kustomize/GB28181/SIP/RTP/SDP/WebRTC/作为上下级域/平台级联互联
Created
2018-05-03
483 commits to develop branch, last one 10 months ago
A Go implementation of STUN
Created
2019-01-21
817 commits to master branch, last one 29 days ago
Linphone is a free VoIP and video softphone based on the SIP protocol. Mirror of linphone-iphone (git://git.linphone.org/linphone-iphone.git)
Created
2015-01-14
5,909 commits to master branch, last one 3 months ago
:books: WebRTC (Web Real-Time Communications) 中文教程
Created
2018-03-24
64 commits to master branch, last one 2 years ago
Generic library for real-time communications with async IO support
Created
2016-09-17
577 commits to master branch, last one 3 months ago
Dockerized FreePBX 15 w/Asterisk 17, Seperate MySQL Database support, and Data Persistence and UCP
This repository has been archived
(exclude archived)
Created
2017-10-01
143 commits to main branch, last one 5 years ago
A fully featured browser based WebRTC SIP phone for Asterisk
Created
2020-03-24
194 commits to master branch, last one 2 months ago
SIP in Go
Created
2017-11-19
457 commits to master branch, last one 2 months ago
The RTPproxy is a high-performance software proxy for RTP streams that can work together with Sippy B2BUA, Kamailio, OpenSIPS and SER.
Created
2014-06-14
4,535 commits to master branch, last one about a month ago
SIP-Based Audit and Attack Tool
Created
2017-09-07
110 commits to master branch, last one 3 years ago
Set of tools to audit SIP based VoIP Systems
Created
2015-05-17
543 commits to master branch, last one 7 days ago