[asterisk-users] voipmonitor.org

[Martin Vit]

Hi,
checkout new open source voipmonitor.org SIP packet sniffer. I've
developed it for my telco company and I've decided to share it.
Testing and contributions are welcome!

VoIPmonitor is open source live network packet sniffer which analyze
SIP and RTP protocol. It can run as daemon or analyzes already
captured pcap files. For each detected VoIP call voipmonitor
calculates statistics about loss, burstiness, latency and predicts MOS
(Meaning Opinion Score) according to ITU-T G.107 E-model. These
statistics are saved to MySQL database and each call is saved as pcap
dump. Web PHP application (it is not part of open source sniffer)
filters data from database and graphs latency and loss distribution.
Voipmonitor also detects improperly terminated calls when BYE or OK
was not seen. To accuratly transform latency to loss packets,
voipmonitor simulates fixed and adaptive jitterbuffer.

Key features

Fast C++ SIP/RTP packet analyzer
Predicts MOS-LQE score according to ITU-T G.107 E-model
Detailed delay/loss statistics stored to MySQL
Each call is saved as standalone pcap file
Jitterbuffer simulator based on asterisk (fixed/adaptive)