Alternatives and similar repositories for janus_gateway:

Users that are interested in janus_gateway are comparing it to the libraries listed below

• agonza1 / WebRTC-Live-Streaming-with-AI
  One RTP video stream to many WebRTC clients with real time image detection using Janus and OpenCV
  ☆27Updated 3 years ago
• OpenTelecom / react-sip-phone
  OTF React SIP.JS Phone
  ☆23Updated last year
• cwysong85 / sipjs-udp
  UDP implementation of the SIP.js library
  ☆37Updated last year
• havfo / Skype-Kamailio-PSTN-gateway
  Skype for Business/Lync PSTN gateway
  ☆24Updated 6 years ago
• benasse / sparrow
  Unofficial armhf build of wazo-platform
  ☆10Updated 4 years ago
• daimoc / Kurento-SIP-GW
  ☆35Updated 3 years ago
• drachtio / drachtio-siprec-recording-server
  SIPREC recording server based on drachtio and rtpengine
  ☆89Updated 9 months ago
• 2600hz / kazoo-configs-kamailio
  ☆35Updated 3 weeks ago
• havfo / Kurento-Nodejs-SIP
  ☆27Updated 7 years ago
• AGProjects / python3-sipsimple
  SIP SIMPLE SDK written in Python
  ☆40Updated last month
• havfo / SipCaller
  React SIP user agent
  ☆53Updated 5 years ago
• webrtcHacks / jitsiLocalRecorder
  100% browser-based, local audio & screen recording for Jitsi Meet
  ☆78Updated 3 years ago
• AGProjects / sylkrtc.js
  JavaScript SylkRTC API library
  ☆29Updated 3 months ago
• ambianic / peerjs-python
  Python port of PeerJS client
  ☆92Updated last year
• Dirvann / video-conference-stitcher
  This package makes it possible to combine separately recorder media files from a video conference call into a single file.
  ☆23Updated last year
• hnimminh / sippscenario
  SIPp Call Scenario for Performance Test
  ☆16Updated 4 years ago
• os11k / dispatcher
  Simple config file of Kamailio as Loadblancer for calls and registrations
  ☆30Updated 3 years ago
• Kurento / mediasoup-demos
  Demos that show how to use mediasoup
  ☆111Updated 2 years ago
• krull / docker-janus
  Debian based docker image for https://github.com/meetecho/janus-gateway
  ☆60Updated last year
• alyssaong1 / VoIPBot
  Bot on an SIP client for VoIP
  ☆117Updated 3 years ago
• drachtio / drachtio-fsmrf
  Drachtio freeswitch-based media resource function -- http://davehorton.github.io/drachtio-fsmrf
  ☆51Updated last month
• meetecho / janode
  A Node.js adapter for the Janus WebRTC server
  ☆104Updated 2 weeks ago
• phsultan / asterisk-eagi-google-speech-recognition
  An example of how to use Asterisk EAGI along with Google Speech recognition to transcribe voice to text
  ☆32Updated 7 years ago
• drachtio / drachtio-rtpengine-webrtcproxy
  Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)
  ☆41Updated last year
• linagora / docker-janus-gateway
  Run janus gateway well configure for hublin in a Docker container.
  ☆66Updated 4 years ago
• fredposner / scripts
  Scripts and such for Kamailio, Asterisk, FreeSWITCH, and more
  ☆39Updated last year
• jambonz / jambonz-feature-server
  Core telephony feature server for the jambones platform
  ☆58Updated this week
• irontec / itsbc
  Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration
  ☆36Updated 6 years ago
• OpenJarbas / baresipy
  baresip python wrapper
  ☆52Updated last month
• sipcapture / hepgen.js
  Barebone HEP Generator for SIP-less Devs
  ☆9Updated 6 months ago