voxserv / kamailio-static-relayLinks
A simple SIP and RTP relay configuration with static routing
☆19Updated 10 years ago
Alternatives and similar repositories for kamailio-static-relay
Users that are interested in kamailio-static-relay are comparing it to the libraries listed below
Sorting:
- Minimalistic FreeSWITCH configuration as a start for new projects☆82Updated 8 years ago
 - FreeSWITCH dialer program for VoIP performance tests☆47Updated 8 years ago
 - SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆169Updated last month
 - Kamailio SIP server docker image☆24Updated 10 years ago
 - Kamailio scripts to call from websocket UA to classic UA, and vice versa.☆43Updated 11 years ago
 - Scripting toolkit for FreeSWITCH written in the Lua programming language☆47Updated last year
 - RTP Cluster is a front-end for multiple RTPproxies☆41Updated last year
 - Kamailio configuration for SIP front-end proxy☆21Updated 13 years ago
 - This repository contains Ansible playbooks and related files for an Active-Passive Kamailio auto-deployment using Pacemaker and Corosync.☆57Updated 6 years ago
 - FreeSWITCH mod_amd☆42Updated last year
 - This Project will provide the inbound sip using that we can route did call to customer ip or customer Phone number☆24Updated 8 years ago
 - 100% Open-Source Packet Capture Agent for HEP☆175Updated 6 months ago
 - Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆35Updated 9 years ago
 - Project to monitor FreeSwitch status and health☆44Updated 8 years ago
 - Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆81Updated 10 years ago
 - Meta-repository to test interop between latest versions of opensips, kamailio, sippy b2bua and rtpproxy☆42Updated last week
 - Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.☆192Updated last week
 - Call Analytics Solution for Freeswitch, Asterisk, Kamailio and other VoIP Switches☆304Updated 3 years ago
 - Create SIP load test scenarios the easy way☆225Updated 5 years ago
 - Load-balancing SIP proxy for Freeswitch☆26Updated 9 years ago
 - Watches etcd keys to update a kamailio dispatcher.list☆21Updated 10 years ago
 - ☆36Updated this week
 - SIP Express Media Server, very fast and flexible SIP media server☆68Updated 2 weeks ago
 - HOMER 5 Docker Containers (OBSOLETE)☆40Updated 4 years ago
 - HOMER 5: Back-End (API) DEPRICATED - use sipcapture/homer-app☆27Updated 5 years ago
 - VoIP signaling and media test automation☆122Updated 2 weeks ago
 - A docker SIPWise rtpengine first-class citizen implementation☆30Updated 6 years ago
 - Dockerfiles to easily build kamailio on different Debian releases☆14Updated 9 years ago
 - Federated SIP deployment☆36Updated 2 years ago
 - Avoided SIP and RTP attackers in Asterisk, FreeSwitch and OpenSIPS☆13Updated 10 years ago