etamme / federated-sipLinks
Federated SIP deployment
☆35Updated last year
Alternatives and similar repositories for federated-sip
Users that are interested in federated-sip are comparing it to the libraries listed below
Sorting:
- SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆168Updated this week
- Barebone Opensource Powered SBC☆109Updated 9 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
- FreeSWITCH dialer program for VoIP performance tests☆46Updated 7 years ago
- Minimalistic FreeSWITCH configuration as a start for new projects☆82Updated 8 years ago
- VoIP signaling and media test automation☆119Updated last month
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆121Updated 3 years ago
- SIPssert Testing Framework is a tool used for facilitating conformity testing of complex VoIP setups☆60Updated 2 months ago
- Scripts and such for Kamailio, Asterisk, FreeSWITCH, and more☆39Updated last year
- SIP performance test tool☆39Updated 2 years ago
- 100% Open-Source Packet Capture Agent for HEP☆172Updated last month
- Create SIP load test scenarios the easy way☆223Updated 4 years ago
- SIP Express Media Server, very fast and flexible SIP media server☆66Updated this week
- async FreeSWITCH cluster control☆74Updated last year
- Kamailio Command Line Interface Control Tool☆53Updated 4 months ago
- ☆35Updated this week
- Kamailio Tutorial Examples from Blog @ https://nickvsnetworking.com/category/voip/kamailio/☆112Updated 2 years ago
- Meta-repository to test interop between latest versions of opensips, kamailio, sippy b2bua and rtpproxy☆43Updated last week
- Sip Express Media Server☆168Updated last month
- UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services☆209Updated last week
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆36Updated 6 years ago
- REST API for sharing IP addresses sending unwanted SIP traffic☆61Updated last year
- 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
- Nagios Plugin to check Call Quality in SIP VoIP (compatible checkmk, etc)☆31Updated 4 years ago
- Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.☆44Updated 3 years ago
- HOMER 7 Docker Images☆96Updated 9 months ago
- Secure SIP Identity Extensions - IETF STIR/SHAKEN - CLI/REST API tool and C library☆50Updated 5 months ago
- Voip Open Linear Testing Suite☆42Updated last week
- Asterisk 13 transcoding module: AMR-WB☆37Updated 3 years ago
- HEP Capture Server for HOMER☆196Updated 2 weeks ago