shimaore / esl
Node.js client and server for FreeSwitch Event Socket
☆131Updated 2 months ago
Alternatives and similar repositories for esl:
Users that are interested in esl are comparing it to the libraries listed below
- FreeSWITCH ESL implementation for Node.js; implements the full Event Socket Library specified in: http://wiki.freeswitch.org/wiki/Esl☆171Updated last year
- A SIP call processing server that can be controlled via nodejs applications☆258Updated this week
- SIPREC recording server based on drachtio and rtpengine☆87Updated 7 months ago
- drachtio signaling resource framework☆173Updated last month
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆115Updated 2 years ago
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆285Updated last year
- Sip Express Media Server☆165Updated 2 months ago
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆162Updated 7 months ago
- FreeSWITCH G.729 module using the opensource bcg729 implementation by Belledonne Communications☆171Updated 3 years ago
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆36Updated 6 years ago
- WebRTC SIP based VoIP client software (+chrome extension)☆103Updated 3 months ago
- sample sipp scenarios for testing freeswitch☆30Updated 2 years ago
- Barebone Opensource Powered SBC☆108Updated 5 months ago
- DEPRECATED!! please use https://github.com/davehorton/drachtio-srf☆69Updated 6 years ago
- ☆37Updated last year
- Drachtio freeswitch-based media resource function -- http://davehorton.github.io/drachtio-fsmrf☆49Updated last month
- Dockerfile for freeswitch☆52Updated 3 years ago
- Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)☆40Updated last year
- sample configurations and settings for opensips for various use-cases☆60Updated 5 years ago
- VoIP signaling and media test automation☆117Updated 2 months ago
- ☆26Updated 7 years ago
- New tryit-jssip application☆88Updated last year
- OpenSIPS + RTPEngine Recording + Speech Recognition in HEP☆21Updated 7 months ago
- FreeSWITCH module to stream audio to websocket and receive response☆88Updated 3 weeks ago
- Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.☆176Updated 2 weeks ago
- OpenSource Freeswitch & Kamailio Billing, rating and Routing Software☆106Updated last year
- A docker SIPWise rtpengine first-class citizen implementation☆28Updated 6 years ago
- UniMRCP modules for Asterisk☆48Updated 7 months ago
- FreeSWITCH mod_xml_curl base configuration classes☆24Updated 8 years ago
- Simple config file of Kamailio as Loadblancer for calls and registrations☆30Updated 3 years ago