unispeech / asterisk-unimrcp
UniMRCP modules for Asterisk
☆48Updated 2 months ago
Related projects: ⓘ
- ☆37Updated 4 years ago
- SIPREC recording server based on drachtio and rtpengine☆81Updated 2 months ago
- A Web Control Panel Application for the OpenSIPS, which is intended for both system and user provisioning. It features more than 18 tools…☆112Updated last month
- ☆61Updated 2 years ago
- A collection of open-sourced freeswitch modules that I use in various drachtio applications☆172Updated 6 months ago
- Simple config file of Kamailio as Loadblancer for calls and registrations☆28Updated 2 years ago
- Speech Recognition in Asterisk with Vosk Server☆102Updated 2 months ago
- Sip Express Media Server☆159Updated 4 months ago
- ☆34Updated last week
- Web Admin Interface for Kamailio☆98Updated 2 months ago
- Web status monitor for FreeSWITCH's mod_callcenter queues and agents☆38Updated 10 years ago
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆36Updated 6 years ago
- OpenSIPS CLI tool - an interactive command line tool that can be used to control and monitor OpenSIPS servers.☆85Updated 2 weeks 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…☆51Updated 11 months ago
- OpenSource G711, G722, G729, Opus & Other Format VoIP SIP Recorder☆151Updated last month
- Kamailio Command Line Interface Control Tool☆51Updated 3 weeks ago
- Simple bidirectional audio protocol☆73Updated 5 months ago
- RTP Cluster is a front-end for multiple RTPproxies☆39Updated 4 months ago
- ☆59Updated 3 months ago
- Linphone.org mirror for bcg729 (git://git.linphone.org/bcg729.git)☆115Updated 3 months ago
- RTP Audio Juicer☆23Updated 2 years ago
- Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.☆44Updated 2 years ago
- Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.☆170Updated 3 weeks ago
- HEP-EEP: Extensible Encapsulation Protocol (Specs & Technical Docs)☆45Updated last year
- FreeSWITCH G.729 module using the opensource bcg729 implementation by Belledonne Communications☆169Updated 2 years ago
- WebRTC SIP based VoIP client software (+chrome extension)☆98Updated 4 years ago
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆106Updated 2 years ago
- Freeswitch Kafka Plugin☆37Updated last year
- sample sipp scenarios for testing freeswitch☆29Updated 2 years ago
- Stream Asterisk audio over Websockets☆155Updated last month