the open source SIP TelePresence system
☆147Dec 10, 2019Updated 6 years ago
Alternatives and similar repositories for telepresence
Users that are interested in telepresence are comparing it to the libraries listed below
Sorting:
- Doubango VoIP framework☆409Jul 20, 2019Updated 6 years ago
- Smart SIP and Media Gateway to connect WebRTC endpoints to any SIP-legacy network☆358Mar 11, 2020Updated 5 years ago
- High Quality Video SIP/IMS client for Google Android☆307Oct 24, 2019Updated 6 years ago
- openmcu.ru☆239Aug 24, 2022Updated 3 years ago
- The world's first HTML5 SIP client (WebRTC)☆956Dec 18, 2020Updated 5 years ago
- OpenVCX (Open Video Conferencing Exchange) is a SIP based video streaming server for multi-way video chat, live web-casting, recording, a…☆66Jan 17, 2019Updated 7 years ago
- SIP WebRTC click-to-call service☆37Mar 4, 2016Updated 9 years ago
- Automatically exported from code.google.com/p/webrtc4all☆10Aug 21, 2015Updated 10 years ago
- g729 codec from: http://g729.googlecode.com/svn/trunk/☆10Nov 13, 2014Updated 11 years ago
- a gb28181 sip sdk based on resiprocate☆30Aug 24, 2018Updated 7 years ago
- OverSIP: the SIP framework you dreamed about☆338Aug 14, 2020Updated 5 years ago
- Unofficial WebRTC Mirror☆163Sep 13, 2017Updated 8 years ago
- Repository of code using JsSIP☆14Oct 24, 2012Updated 13 years ago
- C++ SDP library with ABNF strict parsing☆42Aug 23, 2022Updated 3 years ago
- Utility to test how network losses affects speech quality in VoIP-based applications☆24Jul 18, 2013Updated 12 years ago
- SIP-Client for Android (based on CSipSimple)☆28Aug 31, 2015Updated 10 years ago
- Multiparty-meeting (mediasoup) SIP gateway using Kurento☆28Apr 4, 2019Updated 6 years ago
- SIP Express Media Server, very fast and flexible SIP media server☆70Feb 18, 2026Updated last week
- Web Audio and P2P Call for client side audio conferencing☆14Mar 29, 2016Updated 9 years ago
- C++ SIP stack☆45Dec 13, 2023Updated 2 years ago
- Open Source Communication Provider based on WebRTC and Cloud technologies☆3,141Updated this week
- Voice Broadcasting Software used for marketing, lead generation, political surveys, debt collection and emergency notifications☆176Jan 6, 2022Updated 4 years ago
- TESTING - replicated hourly from Google Code SVN using sync2git☆58Oct 28, 2015Updated 10 years ago
- Linphone.org mirror for linphone-cmake-builder (git://git.linphone.org/linphone-cmake-builder.git)☆12Sep 19, 2023Updated 2 years ago
- Asterisk ARI interface bindings for modern C++☆34Jan 27, 2025Updated last year
- [ARCHIVED] Contents migrated to monorepo: https://github.com/Kurento/kurento☆298Jan 19, 2023Updated 3 years ago
- OfficeSIP Server☆86Sep 5, 2018Updated 7 years ago
- The Sipwise media proxy for Kamailio☆912Feb 23, 2026Updated last week
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆308Oct 19, 2023Updated 2 years ago
- C++ implementation of SIP, ICE, TURN and related protocols.☆769Feb 20, 2026Updated last week
- CSipSimple Mirror (no pull-requests here)☆310Mar 11, 2016Updated 9 years ago
- Web based conference manager for Asterisk & Freeswitch☆25Mar 15, 2011Updated 14 years ago
- A set of build scripts useful for building WebRTC libraries for Android and iOS.☆1,128Apr 26, 2018Updated 7 years ago
- This project can be used to deploy a FreeSWITCH server inside a Docker container. The container currently uses the latest stable release …☆252Apr 10, 2017Updated 8 years ago
- VoIP ID is a Free and Open Source Software to build a hosted VoIP server or hosted IPPBX service. The project is just started, we hope t…☆24Apr 2, 2020Updated 5 years ago
- licode Client for Android☆21Jul 7, 2017Updated 8 years ago
- nodejs client for accessing rtpengine via ng protocol☆25Jul 18, 2023Updated 2 years ago
- the webrtc client for the janus webrtc gateway☆36Feb 19, 2019Updated 7 years ago
- sos(smart os)是基于fslib调试框架开发的一套去中 心化的集rtmp,rtsp,hls,gb28181采集和rtmp,rtsp,hls,gb28181服务器于一体的高性能流媒体服务器,同时也是一款支持rtmp,rtsp,hls,gb28181的NVR,也是一款视…☆35Feb 26, 2023Updated 3 years ago