Alternatives and similar repositories for dana-the-stream-gatekeeper

Users that are interested in dana-the-stream-gatekeeper are comparing it to the libraries listed below

• asterisk / cyber_mega_phone_2k
  ☆62Updated 2 years ago
• retel-io / ari-proxy
  Ari-proxy connects Asterisk, an open source communication server, to the Apache Kafka distributed streaming platform.
  ☆60Updated last week
• havfo / SipCaller
  React SIP user agent
  ☆53Updated 5 years ago
• gmaruzz / saraphone
  SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…
  ☆169Updated 10 months ago
• drachtio / drachtio-rtpengine-webrtcproxy
  Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)
  ☆41Updated last year
• asterisk / asterisk-external-media
  ☆79Updated 2 months ago
• drachtio / drachtio-fsmrf
  Drachtio freeswitch-based media resource function -- http://davehorton.github.io/drachtio-fsmrf
  ☆52Updated this week
• 2600hz / kazoo-configs-kamailio
  ☆35Updated last month
• drachtio / drachtio-siprec-recording-server
  SIPREC recording server based on drachtio and rtpengine
  ☆89Updated 10 months ago
• altanai / kamailioexamples
  VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine
  ☆120Updated 3 years ago
• nimbleape / dana-tsg-ari-bridge
  An ARI Bridge that enables spying on individual participants for STT purposes
  ☆15Updated 2 years ago
• open-voip-alliance / WebphoneLib
  Easier web calling by providing a layer of abstraction around SIP.js
  ☆64Updated last year
• os11k / dispatcher
  Simple config file of Kamailio as Loadblancer for calls and registrations
  ☆32Updated 3 years ago
• CyCoreSystems / ari-proxy
  NATS or RabbitMQ message bus Asterisk REST Interface proxy system implemented in Go
  ☆83Updated 11 months ago
• vicimikec / ViciPhone
  A WebRTC Phone For Vicidial
  ☆32Updated 2 years 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
• irontec / itsbc
  Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration
  ☆36Updated 6 years ago
• cwysong85 / sipjs-udp
  UDP implementation of the SIP.js library
  ☆37Updated last year
• igorolhovskiy / kamailio-dev-dc
  ☆19Updated last year
• staskobzar / amiws
  Asterisk Management Interface (AMI) to Web-socket proxy
  ☆86Updated last year
• nadirhamid / asterisk-audiofork
  Stream Asterisk audio over Websockets
  ☆176Updated 8 months ago
• ricardojlrufino / webphone-sip
  WebRTC SIP based VoIP client software (+chrome extension)
  ☆106Updated 6 months ago
• jchavanton / voip_patrol
  VoIP signaling and media test automation
  ☆119Updated 2 weeks ago
• drachtio / drachtio-srf
  drachtio signaling resource framework
  ☆182Updated this week
• voiceboys / sbcOS
  Barebone Opensource Powered SBC
  ☆108Updated 8 months ago
• minoruta / ast_mongo
  Realtime configuration engine and CDR&CEL backend for Asterisk with MongoDB
  ☆36Updated 6 years ago
• Viking-VoIP / full-voip-on-k8s
  Complete VoIP Solution on kubernetes based on Kamailio/freeSWITCH
  ☆53Updated 2 years ago
• lmangani / docker-rtpengine-speech
  OpenSIPS + RTPEngine Recording + Speech Recognition in HEP
  ☆21Updated 10 months ago
• pascomnet / asterisk_sbr
  Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.
  ☆44Updated 3 years ago
• CyCoreSystems / asterisk-config
  Kubernetes dynamic configuration engine for Asterisk
  ☆67Updated 2 years ago