Alternatives and similar repositories for Kurento-SIP-GW

Users that are interested in Kurento-SIP-GW are comparing it to the libraries listed below

• drachtio / drachtio-rtpengine-webrtcproxy
  Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)
  ☆44Updated 2 years ago
• havfo / Kurento-Nodejs-SIP
  ☆27Updated 7 years ago
• havfo / WEBRTC-to-SIP
  Setup for a WEBRTC client and Kamailio server to call SIP clients
  ☆298Updated last year
• havfo / multiparty-meeting-sipgw
  Multiparty-meeting (mediasoup) SIP gateway using Kurento
  ☆28Updated 6 years ago
• sems-server / sems
  Sip Express Media Server
  ☆172Updated 3 months ago
• havfo / SipCaller
  React SIP user agent
  ☆55Updated 5 years ago
• drachtio / drachtio-siprec-recording-server
  SIPREC recording server based on drachtio and rtpengine
  ☆94Updated last year
• altanai / kamailioexamples
  VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine
  ☆128Updated 3 years ago
• irontec / itsbc
  Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration
  ☆36Updated 7 years ago
• drachtio / drachtio-fsmrf
  Drachtio freeswitch-based media resource function -- http://davehorton.github.io/drachtio-fsmrf
  ☆54Updated 3 weeks ago
• versatica / tryit-jssip
  tryit-jssip application
  ☆88Updated last week
• asterisk / cyber_mega_phone_2k
  ☆63Updated 3 years ago
• sippy / rtp_cluster
  RTP Cluster is a front-end for multiple RTPproxies
  ☆41Updated last year
• ricardojlrufino / webphone-sip
  WebRTC SIP based VoIP client software (+chrome extension)
  ☆108Updated 9 months ago
• sipwise / sems
  SIP Express Media Server, very fast and flexible SIP media server
  ☆68Updated this week
• AGProjects / mediaproxy
  Media relay for RTP/RTCP and UDP streams
  ☆47Updated 4 months ago
• drachtio / drachtio-server
  A SIP call processing server that can be controlled via nodejs applications
  ☆286Updated last month
• nimbleape / dana-the-stream-gatekeeper
  React based front-end demo for Asterisk's SFU
  ☆30Updated 2 years ago
• gmaruzz / saraphone
  SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…
  ☆175Updated last year
• openvoips / OV500
  OpenSource Freeswitch & Kamailio Billing, rating and Routing Software
  ☆112Updated last year
• sippy / b2bua
  Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.
  ☆190Updated 2 months ago
• giavac / janus_freeswitch_integration
  Integrating Janus and FreeSWITCH for an heterogeneous conferencing system
  ☆22Updated 5 years ago
• drachtio / rtpengine-client
  nodejs client for accessing rtpengine via ng protocol
  ☆23Updated 2 years ago
• os11k / dispatcher
  Simple config file of Kamailio as Loadblancer for calls and registrations
  ☆32Updated 3 years ago
• ianblenke / docker-rtpengine
  A docker SIPWise rtpengine first-class citizen implementation
  ☆30Updated 6 years ago
• caruizdiaz / kamailio-ws
  Kamailio scripts to call from websocket UA to classic UA, and vice versa.
  ☆42Updated 11 years ago
• alepolidori / janus-webrtc-phone
  SIP Phone WebRTC for your browser
  ☆60Updated 4 years ago
• os11k / sipp2freeswitch
  sample sipp scenarios for testing freeswitch
  ☆33Updated 3 years ago
• jchavanton / voip_patrol
  VoIP signaling and media test automation
  ☆119Updated last month
• DoubangoTelecom / webrtc2sip
  Smart SIP and Media Gateway to connect WebRTC endpoints to any SIP-legacy network
  ☆354Updated 5 years ago