voxserv / voip_qos_probeLinks
Voice quality probe for end-to-end measurement of jitter and packet loss
☆41Updated 11 years ago
Alternatives and similar repositories for voip_qos_probe
Users that are interested in voip_qos_probe are comparing it to the libraries listed below
Sorting:
- SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆169Updated last month
- Web application to faciliate benchmarking and testing SIP based services☆19Updated 9 years ago
- SIP Voice Quality reports Collector☆15Updated 9 years ago
- Blox route configuration (opensips script)☆22Updated 5 years ago
- Scripting toolkit for FreeSWITCH written in the Lua programming language☆47Updated last year
- Asterisk module that provides the "Flite" dialplan application, which allows you to use the Flite text to speech engine with Asterisk..Wo…☆28Updated 2 years ago
- Minimalistic FreeSWITCH configuration as a start for new projects☆82Updated 8 years ago
- HOMER 5: Back-End (API) DEPRICATED - use sipcapture/homer-app☆27Updated 5 years ago
- RTP Cluster is a front-end for multiple RTPproxies☆41Updated last year
- 100% Open-Source Packet Capture Agent for HEP☆175Updated 6 months ago
- Project to monitor FreeSwitch status and health☆43Updated 8 years ago
- FreeSWITCH dialer program for VoIP performance tests☆47Updated 8 years ago
- RTP Audio Juicer☆25Updated 3 years ago
- async FreeSWITCH cluster control☆74Updated last year
- HEP-EEP: Extensible Encapsulation Protocol (Specs & Technical Docs)☆49Updated 3 months ago
- Restcomm Session Border Controller☆24Updated 6 years ago
- Configurations for Kamailio, FreeSWITCH and all other components of KAZOO☆19Updated 8 years ago
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆35Updated 9 years ago
- Kamailio SIP server docker image☆24Updated 10 years ago
- Create SIP load test scenarios the easy way☆225Updated 5 years ago
- VoIPmonitor sniffer sources☆265Updated this week
- Call Analytics Solution for Freeswitch, Asterisk, Kamailio and other VoIP Switches☆304Updated 3 years ago
- Kamailio configuration for SIP front-end proxy☆21Updated 12 years ago
- Dockerfiles to easily build kamailio on different Debian releases☆14Updated 8 years ago
- HOMER 5 Docker Containers (OBSOLETE)☆40Updated 4 years ago
- SIP Express Media Server, very fast and flexible SIP media server☆68Updated last week
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆81Updated 10 years ago
- Media relay for RTP/RTCP and UDP streams☆47Updated 6 months ago
- Load-balancing SIP proxy for Freeswitch☆26Updated 9 years ago
- Vagrant/Salt configuration for automatically deploying a FreeSWITCH server.☆12Updated 6 years ago