asterisk / asterisk-external-mediaLinks
☆83Updated 4 months ago
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☆181Updated 10 months ago
- A SIP call processing server that can be controlled via nodejs applications☆283Updated this week
- Asterisk Management Interface (AMI) to Web-socket proxy☆87Updated 2 years ago
- SIPREC recording server based on drachtio and rtpengine☆92Updated last year
- UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services☆211Updated last week
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆125Updated 3 years ago
- Kamailio Tutorial Examples from Blog @ https://nickvsnetworking.com/category/voip/kamailio/☆114Updated 2 years ago
- Speech Recognition in Asterisk with Vosk Server☆118Updated last year
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆172Updated last year
- An AudioServer that takes audio from Asterisk via UDP and sends it to Google's Speech To Text Engine☆33Updated 2 years ago
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆295Updated last year
- Barebone Opensource Powered SBC☆109Updated 10 months ago
- Voip Open Linear Testing Suite☆43Updated this week
- ☆19Updated last year
- Node.js client for ARI. This library is best effort with limited support.☆271Updated last year
- Ari-proxy connects Asterisk, an open source communication server, to the Apache Kafka distributed streaming platform.☆61Updated last week
- FreeSWITCH module to stream audio to websocket and receive response☆128Updated last month
- ☆35Updated last month
- Simple config file of Kamailio as Loadblancer for calls and registrations☆32Updated 3 years ago
- Scripts and such for Kamailio, Asterisk, FreeSWITCH, and more☆39Updated last year
- Simple bidirectional audio protocol☆88Updated last month
- OpenSIPS AI Voice Connector Community Edition Platform☆50Updated last week
- Demo of scalable Asterisk on Kubernetes☆172Updated last year
- VoIP signaling and media test automation☆119Updated last week
- Drachtio freeswitch-based media resource function -- http://davehorton.github.io/drachtio-fsmrf☆54Updated last week
- ☆62Updated 3 years ago
- Dockerfile for rtpengine☆24Updated 5 months ago
- Asterisk Queues Dashboard with amiws☆54Updated last year
- Node.js client and server for FreeSwitch Event Socket☆137Updated 7 months ago
- SIPssert Testing Framework is a tool used for facilitating conformity testing of complex VoIP setups☆63Updated 3 months ago