RestComm / Restcomm-ConnectLinks
The Open Source Cloud Communications Platform
☆250Updated 2 years ago
Alternatives and similar repositories for Restcomm-Connect
Users that are interested in Restcomm-Connect are comparing it to the libraries listed below
Sorting:
- RMS - Restcomm Media Server for Real Time Cloud Communications☆168Updated last year
- RestComm SMS Gateway (SMSC) to send/receive SMS from/to Operators Network (GSM)☆137Updated last year
- Call Analytics Solution for Freeswitch, Asterisk, Kamailio and other VoIP Switches☆303Updated 2 years ago
- Leading SIP - IMS - WebRTC Application Server☆245Updated last year
- SIP/XMPP/WebRTC Application Server☆206Updated 5 months ago
- P-KISS-SBC - simple and stupid SIP/RTP SBC - AGPL v3 - Based on kamailio / RTP Engine☆104Updated 3 months ago
- Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://inextrix.atlassian.net/jira/projects☆163Updated last month
- A2Billing is a telecom switch and billing system capable of providing and billing a range of telecom products and services to customers s…☆187Updated 2 years ago
- Javascript library to build a web-broswer softphone☆99Updated last month
- Project to monitor FreeSwitch status and health☆42Updated 8 years ago
- sipstack.io☆71Updated 4 years ago
- RestComm Converged (SIP/HTTP/WebSockets/SMPP) Load Balancer☆27Updated last year
- Free and convenient server process for routing SMS text messages between your applications and SMPP gateways. Interacts with your applica…☆53Updated 9 years ago
- Smart SIP and Media Gateway to connect WebRTC endpoints to any SIP-legacy network☆352Updated 5 years ago
- First Open Source Billing Platform for FreeSWITCH☆49Updated 11 years ago
- Kamailio scripts to call from websocket UA to classic UA, and vice versa.☆41Updated 10 years ago
- SIP WebRTC click-to-call service☆35Updated 9 years ago
- Web Admin Interface for Kamailio☆103Updated 4 months ago
- OverSIP: the SIP framework you dreamed about☆340Updated 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.☆444Updated 3 weeks ago
- Speech recognition script for Asterisk that uses google's speech engine.☆251Updated 7 years ago
- Disclaimer: This repository is a git-svn mirror of the project found at http://java.net/projects/jsip whose original repository is develo…☆160Updated last year
- the open source SIP TelePresence system☆149Updated 5 years ago
- Asterisk ARI interface bindings for Java☆95Updated 9 months ago
- TESTING - replicated hourly from Google Code SVN using sync2git☆59Updated 9 years ago
- WebRTC SIP based VoIP client software (+chrome extension)☆107Updated 7 months ago
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆295Updated last year
- Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.☆188Updated last month
- SIP Express Media Server, very fast and flexible SIP media server☆67Updated this week
- A Web Control Panel Application for the OpenSIPS, which is intended for both system and user provisioning. It features more than 18 tools…☆125Updated 2 weeks ago