Alternatives and similar repositories for audiosocket

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

• nadirhamid / asterisk-audiofork
  Stream Asterisk audio over Websockets
  ☆182Updated 11 months ago
• asterisk / asterisk-external-media
  ☆83Updated 4 months ago
• CyCoreSystems / asterisk-config
  Kubernetes dynamic configuration engine for Asterisk
  ☆68Updated 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…
  ☆54Updated last year
• alphacep / vosk-asterisk
  Speech Recognition in Asterisk with Vosk Server
  ☆119Updated last year
• CyCoreSystems / asterisk-k8s-demo
  Demo of scalable Asterisk on Kubernetes
  ☆172Updated 2 years ago
• ederwander / Asterisk-Google-Speech-Recognition
  ☆37Updated 5 years ago
• sippy / go-b2bua
  The GO port of the Sippy B2BUA
  ☆68Updated 3 months ago
• CyCoreSystems / ari-proxy
  NATS or RabbitMQ message bus Asterisk REST Interface proxy system implemented in Go
  ☆84Updated last year
• Viking-VoIP / full-voip-on-k8s
  Complete VoIP Solution on kubernetes based on Kamailio/freeSWITCH
  ☆54Updated 2 years ago
• drachtio / drachtio-siprec-recording-server
  SIPREC recording server based on drachtio and rtpengine
  ☆94Updated last year
• akam-it / kamailio-ha
  Kamailio+rtpengine failover demo infrastructure for kamailio-ru-2020 conference
  ☆15Updated 4 years ago
• silentindark / audiosocket_server
  Asterisk AudioSocket server in Python
  ☆37Updated 4 years ago
• havfo / WEBRTC-to-SIP
  Setup for a WEBRTC client and Kamailio server to call SIP clients
  ☆297Updated last year
• altanai / kamailioexamples
  VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine
  ☆126Updated 3 years ago
• 2600hz / kazoo-configs-kamailio
  ☆35Updated last week
• hnimminh / sippscenario
  SIPp Call Scenario for Performance Test
  ☆17Updated 4 years ago
• lmangani / docker-rtpengine-speech
  OpenSIPS + RTPEngine Recording + Speech Recognition in HEP
  ☆21Updated last year
• igorolhovskiy / volts
  Voip Open Linear Testing Suite
  ☆43Updated last week
• retel-io / ari-proxy
  Ari-proxy connects Asterisk, an open source communication server, to the Apache Kafka distributed streaming platform.
  ☆61Updated this week
• voipxswitch / kamailio-jsonrpc-client
  ☆24Updated last year
• dOpensource / dsiprouter
  UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services
  ☆211Updated last week
• jchavanton / voip_patrol
  VoIP signaling and media test automation
  ☆119Updated last month
• drachtio / drachtio-server
  A SIP call processing server that can be controlled via nodejs applications
  ☆286Updated 3 weeks ago
• asipto / secsipidx
  Secure SIP Identity Extensions - IETF STIR/SHAKEN - CLI/REST API tool and C library
  ☆50Updated 3 weeks ago
• asterisk / cyber_mega_phone_2k
  ☆63Updated 3 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
• phsultan / asterisk-eagi-google-speech-recognition
  An example of how to use Asterisk EAGI along with Google Speech recognition to transcribe voice to text
  ☆32Updated 7 years ago
• pchero / asterisk-tiresias
  Audio fingerprinting and recognition module for the Asterisk
  ☆10Updated 7 years ago
• miconda / sipexer
  Modern and flexible SIP/VoIP cli tool
  ☆354Updated last month