Alternatives and similar repositories for Kurento-SIP-GW

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

• havfo / WEBRTC-to-SIP
  Setup for a WEBRTC client and Kamailio server to call SIP clients
  ☆308Updated 2 years ago
• drachtio / drachtio-rtpengine-webrtcproxy
  Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)
  ☆44Updated 2 years ago
• havfo / multiparty-meeting-sipgw
  Multiparty-meeting (mediasoup) SIP gateway using Kurento
  ☆28Updated 6 years ago
• havfo / Kurento-Nodejs-SIP
  ☆27Updated 8 years ago
• sems-server / sems
  Sip Express Media Server
  ☆180Updated last month
• drachtio / drachtio-siprec-recording-server
  SIPREC recording server based on drachtio and rtpengine
  ☆95Updated last year
• havfo / SipCaller
  React SIP user agent
  ☆56Updated 6 years ago
• asterisk / cyber_mega_phone_2k
  ☆62Updated 3 years ago
• sippy / b2bua
  Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.
  ☆193Updated last month
• altanai / kamailioexamples
  VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine
  ☆130Updated 3 years ago
• versatica / tryit-jssip
  tryit-jssip application
  ☆88Updated 4 months ago
• drachtio / drachtio-server
  A SIP call processing server that can be controlled via nodejs applications
  ☆297Updated last week
• irontec / itsbc
  Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration
  ☆37Updated 7 years ago
• AGProjects / mediaproxy
  Media relay for RTP/RTCP and UDP streams
  ☆47Updated last month
• voiceip / oreka
  OpenSource G711, G722, G729, Opus & Other Format VoIP SIP Recorder
  ☆179Updated 5 months ago
• os11k / sipp2freeswitch
  sample sipp scenarios for testing freeswitch
  ☆34Updated 3 years ago
• drachtio / drachtio-fsmrf
  Drachtio freeswitch-based media resource function -- http://davehorton.github.io/drachtio-fsmrf
  ☆55Updated 3 months ago
• caruizdiaz / kamailio-ws
  Kamailio scripts to call from websocket UA to classic UA, and vice versa.
  ☆43Updated 11 years ago
• sippy / rtp_cluster
  RTP Cluster is a front-end for multiple RTPproxies
  ☆41Updated last year
• gmaruzz / saraphone
  SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…
  ☆183Updated last year
• sipwise / sems
  SIP Express Media Server, very fast and flexible SIP media server
  ☆70Updated this week
• ricardojlrufino / webphone-sip
  WebRTC SIP based VoIP client software (+chrome extension)
  ☆111Updated last year
• DoubangoTelecom / webrtc2sip
  Smart SIP and Media Gateway to connect WebRTC endpoints to any SIP-legacy network
  ☆357Updated 5 years ago
• os11k / dispatcher
  Simple config file of Kamailio as Loadblancer for calls and registrations
  ☆31Updated 4 years ago
• altanai / opensipsexamples
  sample configurations and settings for opensips for various use-cases
  ☆63Updated 6 years ago
• drachtio / rtpengine-client
  nodejs client for accessing rtpengine via ng protocol
  ☆25Updated 2 years ago
• OpenSIPS / opensips-cp
  A Web Control Panel Application for the OpenSIPS, which is intended for both system and user provisioning. It features more than 18 tools…
  ☆127Updated last month
• openvoips / OV500
  OpenSource Freeswitch & Kamailio Billing, rating and Routing Software
  ☆114Updated last year
• nimbleape / dana-the-stream-gatekeeper
  React based front-end demo for Asterisk's SFU
  ☆30Updated 2 years ago
• jchavanton / voip_patrol
  VoIP signaling and media test automation
  ☆123Updated 2 months ago