[asterisk-users] voipmonitor.org

[Jeff Brower]

Martin-

> 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.

How many channels can it handle simultaneously?  How does it do MOS prediction if low bitrate codecs are being used
(G729, AMR, etc)?

Thanks.

-Jeff

> 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)