voxserv / kamailio-static-relay
A simple SIP and RTP relay configuration with static routing
☆19Updated 9 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
- FreeSWITCH dialer program for VoIP performance tests☆44Updated 7 years ago
- This Project will provide the inbound sip using that we can route did call to customer ip or customer Phone number☆24Updated 7 years ago
- SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆167Updated 10 months ago
- Kamailio SIP server docker image☆24Updated 9 years ago
- Kamailio configuration for SIP front-end proxy☆21Updated 12 years ago
- RTP Cluster is a front-end for multiple RTPproxies☆41Updated 10 months ago
- Voice quality probe for end-to-end measurement of jitter and packet loss☆41Updated 10 years ago
- A curated list of HEP / EEP enabled projects☆28Updated 6 years ago
- FreeSWITCH socket client written in go (http://golang.org)☆72Updated 3 months ago
- SIP Express Media Server, very fast and flexible SIP media server☆62Updated this week
- HOMER 5: Back-End (API) DEPRICATED - use sipcapture/homer-app☆27Updated 5 years ago
- Kamailio scripts to call from websocket UA to classic UA, and vice versa.☆40Updated 10 years ago
- WebSocket Command Line Tool For SIP And Template Data☆42Updated 10 months ago
- This repository contains Ansible playbooks and related files for an Active-Passive Kamailio auto-deployment using Pacemaker and Corosync.☆54Updated 6 years ago
- Minimalistic FreeSWITCH configuration as a start for new projects☆80Updated 8 years ago
- Chef cookbook to install FreeSWITCH☆24Updated 9 years ago
- Asterisk REST Interface proxy system written in Go☆32Updated 5 years ago
- Project to monitor FreeSwitch status and health☆41Updated 7 years ago
- FreeSWITCH mod_amd☆41Updated last year
- Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)☆40Updated last year
- MOS Score Monitoring for FusionPBX/FreeSwitch☆10Updated 9 years ago
- A docker SIPWise rtpengine first-class citizen implementation☆29Updated 6 years ago
- Blox route configuration (opensips script)☆22Updated 4 years ago
- ☆34Updated last week
- Load-balancing SIP proxy for Freeswitch☆24Updated 8 years ago
- sample sipp scenarios for testing freeswitch☆30Updated 2 years ago
- Audio fingerprinting and recognition module for the Asterisk☆10Updated 6 years ago
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆35Updated 8 years ago
- 100% Open-Source Packet Capture Agent for HEP☆172Updated 3 weeks ago
- Pipe arbitrary data rows (logs, events, cdrs, esl, etc) to HEP Server (HOMER)☆25Updated 8 months ago