nimbleape / dana-tsg-ari-bridge
An ARI Bridge that enables spying on individual participants for STT purposes
☆14Updated last year
Related projects ⓘ
Alternatives and complementary repositories for dana-tsg-ari-bridge
- An AudioServer that takes audio from Asterisk via UDP and sends it to Google's Speech To Text Engine☆32Updated last year
- ☆66Updated 5 months ago
- ☆22Updated last year
- ☆43Updated 5 months ago
- Speech Recognition in Asterisk with Vosk Server☆105Updated 4 months ago
- A SIP call processing server that can be controlled via nodejs applications☆247Updated this week
- ☆21Updated 5 years ago
- Astricon 2019 presentation on Asterisk media streaming☆25Updated 5 years ago
- SIPREC recording server based on drachtio and rtpengine☆83Updated 4 months ago
- React based front-end demo for Asterisk's SFU☆30Updated last year
- Stream Asterisk audio over Websockets☆158Updated 2 months ago
- Asterisk speech driver for Google Dialogflow☆16Updated 4 years ago
- Simple bidirectional audio protocol☆77Updated 7 months ago
- drachtio signaling resource framework☆169Updated this week
- Simple config file of Kamailio as Loadblancer for calls and registrations☆28Updated 3 years ago
- ☆37Updated 4 years ago
- ☆34Updated this week
- Barebone Opensource Powered SBC☆106Updated 2 months ago
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆153Updated 4 months ago
- Kubernetes dynamic configuration engine for Asterisk☆64Updated 2 years ago
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆35Updated 6 years ago
- Core telephony feature server for the jambones platform☆46Updated this week
- ☆17Updated 6 months ago
- SIP Phone WebRTC for your browser☆57Updated 3 years ago
- Kamailio+rtpengine failover demo infrastructure for kamailio-ru-2020 conference☆15Updated 3 years ago
- FreeSWITCH module to stream audio to websocket and receive response☆75Updated 3 months ago
- Voip Open Linear Testing Suite☆39Updated last month
- ☆61Updated 2 years ago
- WebRTC SIP based VoIP client software (+chrome extension)☆99Updated 4 years ago