amigniter / mod_audio_stream
FreeSWITCH module to stream audio to websocket and receive response
☆112Updated last month
Alternatives and similar repositories for mod_audio_stream:
Users that are interested in mod_audio_stream are comparing it to the libraries listed below
- sample configurations and settings for opensips for various use-cases☆60Updated 5 years ago
- SIPREC recording server based on drachtio and rtpengine☆89Updated 9 months ago
- FreeSWITCH ASR☆40Updated 4 years ago
- API server and Web GUI for FreeSwitch written in Golang and Angular☆73Updated last month
- A FreeSWITCH module to interface to your speech recognition server over websocket☆33Updated last year
- ☆77Updated last month
- Kamailio Wiki with content in markdown format☆39Updated last week
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆119Updated 3 years ago
- ☆38Updated 2 months ago
- FreeSWITCH ASR module fork from mod_audio_stream, use FunASR online cpu version☆21Updated 4 months ago
- FreeSWITCH G.729 module using the opensource bcg729 implementation by Belledonne Communications☆172Updated 3 years ago
- This is FreeSwitch module that can do VAD and ASR with IFLYTEK websocket api.☆38Updated 2 years ago
- A real time stream trans to other application or server for FreeSWITCH☆34Updated 2 months ago
- TTS and ASR module with auto Voice Active Detecting supported for Freeswitch. I build it for Nature sound interactive, With the embedded…☆69Updated 5 years ago
- Kamailio Tutorial Examples from Blog @ https://nickvsnetworking.com/category/voip/kamailio/☆111Updated 2 years ago
- Barebone Opensource Powered SBC☆108Updated 8 months ago
- SIPssert Testing Framework is a tool used for facilitating conformity testing of complex VoIP setups☆59Updated last month
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆169Updated 10 months ago
- Hacked FreeSWITCH-G729 speech codec using Intel® Integrated Performance Primitive.☆19Updated 12 years ago
- Stream Asterisk audio over Websockets☆176Updated 8 months ago
- Dockerfile for rtpengine☆23Updated 3 months ago
- sip capture server by hep。work with OpenSIPS, Kamailo, and FreeSWITCH。☆66Updated 2 months ago
- OpenSIPS CLI tool - an interactive command line tool that can be used to control and monitor OpenSIPS servers.☆93Updated 5 months ago
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆288Updated last year
- Various SIP resources.☆225Updated 8 months ago
- A SIP call processing server that can be controlled via nodejs applications☆274Updated last week
- OpenSource G711, G722, G729, Opus & Other Format VoIP SIP Recorder☆169Updated 3 months ago
- Learn Kamailio☆29Updated 7 months ago
- FreeSWITCH mod_xml_curl base configuration classes☆24Updated 8 years ago
- VoIP signaling and media test automation☆119Updated last week