Alternatives and similar repositories for asterisk-sipml5:

Users that are interested in asterisk-sipml5 are comparing it to the libraries listed below

• asterisk / cyber_mega_phone_2k
  ☆61Updated 2 years ago
• os11k / dispatcher
  Simple config file of Kamailio as Loadblancer for calls and registrations
  ☆29Updated 3 years ago
• drachtio / drachtio-siprec-recording-server
  SIPREC recording server based on drachtio and rtpengine
  ☆85Updated 6 months ago
• havfo / SipCaller
  React SIP user agent
  ☆52Updated 5 years ago
• daimoc / Kurento-SIP-GW
  ☆35Updated 3 years ago
• 2600hz / libwebphone
  Javascript library to build a web-broswer softphone
  ☆96Updated last month
• 2600hz / kazoo-configs-kamailio
  ☆34Updated 2 months ago
• fredposner / scripts
  Scripts and such for Kamailio, Asterisk, FreeSWITCH, and more
  ☆38Updated last year
• minoruta / ast_mongo
  Realtime configuration engine and CDR&CEL backend for Asterisk with MongoDB
  ☆36Updated 6 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…
  ☆51Updated last year
• altanai / kamailioexamples
  VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine
  ☆114Updated 2 years ago
• asipto / siremis
  Web Admin Interface for Kamailio
  ☆100Updated 3 months ago
• davidcsi / kamailio-private-public
  Kamailio config for public/private proxy with rtpengine
  ☆9Updated last year
• staskobzar / amiws
  Asterisk Management Interface (AMI) to Web-socket proxy
  ☆87Updated last year
• ederwander / Asterisk-Google-Speech-Recognition
  ☆37Updated 4 years ago
• ricardojlrufino / webphone-sip
  WebRTC SIP based VoIP client software (+chrome extension)
  ☆103Updated 2 months ago
• voipxswitch / kamailio-jsonrpc-client
  ☆24Updated last year
• custompbx / custompbx
  API server and Web GUI for FreeSwitch written in Golang and Angular
  ☆62Updated last month
• USAN / res_speech_gdfe
  Asterisk speech driver for Google Dialogflow
  ☆16Updated 4 years ago
• OpenSIPS / opensips-cli
  OpenSIPS CLI tool - an interactive command line tool that can be used to control and monitor OpenSIPS servers.
  ☆90Updated 2 months ago
• nimbleape / dana-tsg-rtp-stt-audioserver
  An AudioServer that takes audio from Asterisk via UDP and sends it to Google's Speech To Text Engine
  ☆33Updated 2 years ago
• nimbleape / dana-the-stream-gatekeeper
  React based front-end demo for Asterisk's SFU
  ☆30Updated 2 years ago
• asterisk / asterisk-external-media
  ☆70Updated 7 months ago
• agilityfeat / webrtc-sip-example
  A small example of how to build a WebRTC application using SIP as signaling layer
  ☆34Updated 5 years ago
• nadirhamid / audiofork-transcribe-demo
  Example setup of Google Cloud speech with Asterisk audio fork
  ☆12Updated 7 months ago
• nimbleape / dana-tsg-ari-bridge
  An ARI Bridge that enables spying on individual participants for STT purposes
  ☆14Updated 2 years ago
• staskobzar / amiws_queue
  Asterisk Queues Dashboard with amiws
  ☆52Updated last year
• drivefast / asterisk-res_json
  asterisk json utilities
  ☆42Updated 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…
  ☆160Updated 7 months ago
• alphacep / vosk-asterisk
  Speech Recognition in Asterisk with Vosk Server
  ☆107Updated 7 months ago