Alternatives and similar repositories for mod_openai_asr

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

• akscf / mod_whisper_asr
  Freeswitch ASR module
  ☆19Updated 9 months ago
• amigniter / mod_audio_stream
  FreeSWITCH module to stream audio to websocket and receive response
  ☆157Updated 2 months ago
• custompbx / custompbx
  API server and Web GUI for FreeSwitch written in Golang and Angular
  ☆95Updated 7 months ago
• drachtio / drachtio-siprec-recording-server
  SIPREC recording server based on drachtio and rtpengine
  ☆95Updated last year
• cyrenity / mod_whisper
  A FreeSWITCH module to interface to your speech recognition server over websocket
  ☆37Updated 4 months ago
• wazo-platform / wazo-c4-sbc
  Wazo Platform SBC Kamailio configuration for the C4 infrastructure
  ☆13Updated 2 years ago
• freeswitch / mod_unimrcp
  ☆50Updated last month
• Slonik / FreeSWITCH-panel
  FreeSWITCH admin panel
  ☆24Updated 13 years ago
• irontec / itsbc
  Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration
  ☆37Updated 7 years ago
• voipbin / monorepo
  voipbin backend monorepo
  ☆29Updated last week
• gmaruzz / saraphone
  SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…
  ☆182Updated last year
• wangduanduan / siphub
  sip capture server by hep。work with OpenSIPS, Kamailo, and FreeSWITCH。
  ☆69Updated 4 months ago
• ricardojlrufino / webphone-sip
  WebRTC SIP based VoIP client software (+chrome extension)
  ☆111Updated 11 months ago
• voipxswitch / kamailio-jsonrpc-client
  ☆24Updated last year
• os11k / dispatcher
  Simple config file of Kamailio as Loadblancer for calls and registrations
  ☆32Updated 4 years ago
• altanai / kamailioexamples
  VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine
  ☆130Updated 3 years ago
• florentchauveau / freeswitch_exporter
  Prometheus exporter for FreeSWITCH
  ☆30Updated 3 years ago
• OpenSIPS / call-api
  Call API is a front-end layer for managing advanced SIP call flows. It listens for WebSocket connections and talks JSON-RPC 2.0 over them…
  ☆55Updated 2 months ago
• davehorton / drachtio-siprec-recording-client
  SIP outbound proxy based on drachtio and freeswitch that includes siprec client functionality
  ☆20Updated last year
• voiceboys / sbcOS
  Barebone Opensource Powered SBC
  ☆111Updated 3 months ago
• hnimminh / sip-healthcheck
  Monitor SIP server and Notify whenever downtime/latency detected.
  ☆15Updated 5 years ago
• freeswitch / verto-client
  ☆38Updated 2 years ago
• hnimminh / sippscenario
  SIPp Call Scenario for Performance Test
  ☆17Updated 5 years ago
• Viking-VoIP / full-voip-on-k8s
  Complete VoIP Solution on kubernetes based on Kamailio/freeSWITCH
  ☆57Updated 2 years ago
• irontec / bbs
  Black Box SIP Tester
  ☆31Updated 11 months ago
• alepolidori / janus-webrtc-phone
  SIP Phone WebRTC for your browser
  ☆59Updated 4 years ago
• fredposner / scripts
  Scripts and such for Kamailio, Asterisk, FreeSWITCH, and more
  ☆39Updated last month
• openvoips / OV500
  OpenSource Freeswitch & Kamailio Billing, rating and Routing Software
  ☆112Updated last year
• EvoluxBR / greenswitch
  Battle proven FreeSWITCH Event Socket Protocol client implementation with Gevent
  ☆144Updated 6 months ago
• mnovicio / ptt-freeswitch-ui
  Bare UI for Push-To-Talk and SIP Call/Messaging
  ☆22Updated 5 years ago