unispeech / asterisk-unimrcpLinks
UniMRCP modules for Asterisk
☆51Updated last year
Alternatives and similar repositories for asterisk-unimrcp
Users that are interested in asterisk-unimrcp are comparing it to the libraries listed below
Sorting:
- ☆37Updated 5 years ago
- ☆63Updated 3 years ago
- A Web Control Panel Application for the OpenSIPS, which is intended for both system and user provisioning. It features more than 18 tools…☆127Updated last week
- Web Admin Interface for Kamailio☆102Updated 9 months ago
- SIPREC recording server based on drachtio and rtpengine☆95Updated last year
- OpenSource Freeswitch & Kamailio Billing, rating and Routing Software☆112Updated last year
- Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.☆192Updated last week
- Sip Express Media Server☆178Updated 3 weeks ago
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆306Updated 2 years ago
- OpenSource G711, G722, G729, Opus & Other Format VoIP SIP Recorder☆177Updated 4 months ago
- Speech Recognition in Asterisk with Vosk Server☆126Updated last year
- WebRTC SIP based VoIP client software (+chrome extension)☆111Updated last year
- Simple config file of Kamailio as Loadblancer for calls and registrations☆32Updated 4 years ago
- Load-balancing SIP proxy for Freeswitch☆27Updated 9 years ago
- React SIP user agent☆57Updated 6 years ago
- Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.☆44Updated 3 years ago
- OpenSIPS CLI tool - an interactive command line tool that can be used to control and monitor OpenSIPS servers.☆98Updated 2 months ago
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆130Updated 3 years ago
- Call API is a front-end layer for managing advanced SIP call flows. It listens for WebSocket connections and talks JSON-RPC 2.0 over them…☆55Updated 2 months ago
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆183Updated last year
- Kamailio scripts to call from websocket UA to classic UA, and vice versa.☆43Updated 11 years ago
- SIP Phone WebRTC for your browser☆59Updated 4 years ago
- Docker Image Repository for OpenSIPS☆82Updated 5 months ago
- Asterisk speech driver for Google Dialogflow☆16Updated 5 years ago
- The RTPproxy is a high-performance software proxy for RTP streams that can work together with Sippy B2BUA, Kamailio, OpenSIPS and SER.☆458Updated 2 weeks ago
- Simple bidirectional audio protocol☆101Updated last month
- SIPp scenarios I use for testing SIP stuff☆300Updated last year
- sample configurations and settings for opensips for various use-cases☆63Updated 6 years ago
- sip capture server by hep。work with OpenSIPS, Kamailo, and FreeSWITCH。☆69Updated 4 months ago
- ☆88Updated 8 months ago