voxserv / kamailio-static-relay
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:
- FreeSWITCH dialer program for VoIP performance tests☆46Updated 7 years ago
- Load-balancing SIP proxy for Freeswitch☆24Updated 8 years ago
- Chef cookbook to install FreeSWITCH☆25Updated 9 years ago
- Kamailio configuration for SIP front-end proxy☆21Updated 12 years ago
- Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)☆41Updated last year
- RTP Cluster is a front-end for multiple RTPproxies☆41Updated last year
- Minimalistic FreeSWITCH configuration as a start for new projects☆82Updated 8 years ago
- This repository contains Ansible playbooks and related files for an Active-Passive Kamailio auto-deployment using Pacemaker and Corosync.☆54Updated 6 years ago
- Kamailio scripts to call from websocket UA to classic UA, and vice versa.☆40Updated 10 years ago
- MOS Score Monitoring for FusionPBX/FreeSwitch☆10Updated 9 years ago
- Kamailio SIP server docker image☆24Updated 10 years ago
- Asterisk REST Interface proxy system written in Go☆32Updated 5 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
- A curated list of HEP / EEP enabled projects☆28Updated 6 years ago
- Meta-repository to test interop between latest versions of opensips, kamailio, sippy b2bua and rtpproxy☆43Updated this week
- SIP Express Media Server, very fast and flexible SIP media server☆65Updated last week
- Dockerfiles to easily build kamailio on different Debian releases☆14Updated 8 years ago
- SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆169Updated 11 months ago
- Scripting toolkit for FreeSWITCH written in the Lua programming language☆47Updated last year
- Project to monitor FreeSwitch status and health☆42Updated 8 years ago
- Blox route configuration (opensips script)☆22Updated 4 years ago
- Voice quality probe for end-to-end measurement of jitter and packet loss☆41Updated 10 years ago
- ☆35Updated last month
- FreeSWITCH mod_amd☆41Updated last year
- FreeSWITCH socket client written in go (http://golang.org)☆73Updated 4 months ago
- WebSocket Command Line Tool For SIP And Template Data☆42Updated 11 months ago
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆79Updated 10 years ago
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆35Updated 8 years ago
- Pipe arbitrary data rows (logs, events, cdrs, esl, etc) to HEP Server (HOMER)☆25Updated 10 months ago
- 100% Open-Source Packet Capture Agent for HEP☆172Updated 3 weeks ago