englercj / node-eslLinks
FreeSWITCH ESL implementation for Node.js; implements the full Event Socket Library specified in: http://wiki.freeswitch.org/wiki/Esl
☆174Updated last year
Alternatives and similar repositories for node-esl
Users that are interested in node-esl are comparing it to the libraries listed below
Sorting:
- Node.js client and server for FreeSwitch Event Socket☆140Updated last year
- A SIP call processing server that can be controlled via nodejs applications☆297Updated this week
- This project can be used to deploy a FreeSWITCH server inside a Docker container. The container currently uses the latest stable release …☆251Updated 8 years ago
- drachtio signaling resource framework☆200Updated last month
- Node.js client for ARI. This library is best effort with limited support.☆274Updated last year
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆307Updated 2 years ago
- Session Initiation Protocol for node.js☆439Updated last year
- Drachtio freeswitch-based media resource function -- http://davehorton.github.io/drachtio-fsmrf☆55Updated 3 months ago
- DEPRECATED!! please use https://github.com/davehorton/drachtio-srf☆69Updated 7 years ago
- SIPREC recording server based on drachtio and rtpengine☆95Updated last year
- NodeJS Asterisk Manager API☆262Updated 3 years ago
- tryit-jssip application☆88Updated 3 months ago
- A pjsip/pjsua2 binding for node.js☆194Updated 9 years ago
- FreeSWITCH G.729 module using the opensource bcg729 implementation by Belledonne Communications☆177Updated 4 months ago
- Load-balancing SIP proxy for Freeswitch☆27Updated 9 years ago
- ctxSip is a Javascript based SIP client that uses WebRTC and WebSockets to connect to your SIP server. The UI is designed to be launched …☆332Updated last year
- Kamailio scripts to call from websocket UA to classic UA, and vice versa.☆43Updated 11 years ago
- Smart SIP and Media Gateway to connect WebRTC endpoints to any SIP-legacy network☆357Updated 5 years ago
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆130Updated 3 years ago
- ☆88Updated last week
- The world's first HTML5 SIP client (WebRTC)☆955Updated 4 years ago
- VoIP signaling and media test automation☆122Updated last month
- Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)☆44Updated 2 years ago
- OpenSource G711, G722, G729, Opus & Other Format VoIP SIP Recorder☆179Updated 5 months ago
- FreeSWITCH mod_amd☆42Updated 2 years ago
- SIPp scenarios I use for testing SIP stuff☆301Updated last year
- Web Admin Interface for Kamailio☆102Updated 9 months ago
- Dockerfile for freeswitch☆51Updated 4 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 last month
- The Sipwise media proxy for Kamailio☆892Updated last week