asterisk / asterisk-external-mediaLinks
☆90Updated last month
Alternatives and similar repositories for asterisk-external-media
Users that are interested in asterisk-external-media are comparing it to the libraries listed below
Sorting:
- Stream Asterisk audio over Websockets☆188Updated 4 months ago
- A SIP call processing server that can be controlled via nodejs applications☆301Updated 2 weeks ago
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆130Updated last week
- Asterisk Management Interface (AMI) to Web-socket proxy☆92Updated 2 years ago
- SIPREC recording server based on drachtio and rtpengine☆95Updated last year
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆183Updated last year
- UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services☆224Updated last week
- Speech Recognition in Asterisk with Vosk Server☆128Updated last year
- Kamailio Tutorial Examples from Blog @ https://nickvsnetworking.com/category/voip/kamailio/☆123Updated 3 years ago
- Simple bidirectional audio protocol☆108Updated 2 months ago
- Barebone Opensource Powered SBC☆114Updated 6 months ago
- An AudioServer that takes audio from Asterisk via UDP and sends it to Google's Speech To Text Engine☆33Updated 3 years ago
- Voip Open Linear Testing Suite☆44Updated 4 months ago
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆308Updated 2 years ago
- Drachtio freeswitch-based media resource function -- http://davehorton.github.io/drachtio-fsmrf☆56Updated 4 months ago
- ☆19Updated 3 weeks ago
- Simple config file of Kamailio as Loadblancer for calls and registrations☆31Updated 4 years ago
- VoIP signaling and media test automation☆123Updated 2 months ago
- Scripts and such for Kamailio, Asterisk, FreeSWITCH, and more☆40Updated 3 months ago
- Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.☆44Updated 3 years ago
- Modern and flexible SIP/VoIP cli tool☆374Updated last week
- ☆36Updated 2 weeks ago
- OpenSource G711, G722, G729, Opus & Other Format VoIP SIP Recorder☆181Updated 3 weeks ago
- Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)☆44Updated 2 years ago
- Ari-proxy connects Asterisk, an open source communication server, to the Apache Kafka distributed streaming platform.☆63Updated last month
- ☆62Updated 3 years ago
- Demo of scalable Asterisk on Kubernetes☆174Updated 2 years ago
- Asterisk external speech to text application☆62Updated 2 years ago
- FreeSWITCH module to stream audio to websocket and receive response☆176Updated 2 weeks ago
- React based front-end demo for Asterisk's SFU☆30Updated 3 years ago