sboily / wazo-hackathon-asterisk-stream-moduleLinks
☆22Updated 6 years ago
Alternatives and similar repositories for wazo-hackathon-asterisk-stream-module
Users that are interested in wazo-hackathon-asterisk-stream-module are comparing it to the libraries listed below
Sorting:
- Astricon 2019 presentation on Asterisk media streaming☆30Updated 6 years ago
- Stream Asterisk audio over Websockets☆186Updated 2 months ago
- An ARI Bridge that enables spying on individual participants for STT purposes☆16Updated 2 years ago
- Speech Recognition in Asterisk with Vosk Server☆126Updated last year
- ☆37Updated 5 years ago
- Asterisk speech driver for Google Dialogflow☆16Updated 5 years ago
- ☆88Updated 8 months ago
- Asterisk AudioSocket server in Python☆41Updated 4 years ago
- An AudioServer that takes audio from Asterisk via UDP and sends it to Google's Speech To Text Engine☆33Updated 2 years ago
- Simple bidirectional audio protocol☆102Updated last month
- FreeSWITCH module to stream audio to websocket and receive response☆163Updated 3 months ago
- Example setup of Google Cloud speech with Asterisk audio fork☆12Updated last year
- ☆24Updated 2 years ago
- Demo of scalable Asterisk on Kubernetes☆173Updated 2 years ago
- API server and Web GUI for FreeSwitch written in Golang and Angular☆96Updated 8 months ago
- A SIP call processing server that can be controlled via nodejs applications☆293Updated 4 months ago
- Minimalist Windows / OSx / Linux SIP Softphone with API Control☆151Updated 8 months ago
- Kamailio Tutorial Examples from Blog @ https://nickvsnetworking.com/category/voip/kamailio/☆122Updated 2 years ago
- Kubernetes dynamic configuration engine for Asterisk☆70Updated 3 years ago
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆130Updated 3 years ago
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆183Updated last year
- Asterisk Streaming Connection with google speech API☆14Updated 9 years ago
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆306Updated 2 years ago
- SIPREC recording server based on drachtio and rtpengine☆95Updated last year
- React based front-end demo for Asterisk's SFU☆30Updated 2 years ago
- Battle proven FreeSWITCH Event Socket Protocol client implementation with Gevent☆145Updated 6 months ago
- Web status monitor for FreeSWITCH's mod_callcenter queues and agents☆39Updated 11 years ago
- Barebone Opensource Powered SBC☆113Updated 4 months ago
- ☆19Updated last year
- This repository contains Ansible playbooks and related files for an Active-Passive Kamailio auto-deployment using Pacemaker and Corosync.☆57Updated 6 years ago