VoIPmonitor 2.1.1 version released

 

VoIPmonitor is open source live network packet sniffer and call recorder for linux or posix unix which analyzes 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 so operator centers can quickly decide which calls are bad. These statistics are saved to MySQL database and each call is saved as pcap dump and optionaly to audio WAV. Free VoIPmonitor can decode only G.711 codec. Commercial version can decode from G.729/G.723/GSM/Speex/iLBC. One of unique feature of VoIPmonitor is decoding phone calls which are changing codecs during call. Synchronisation is achieved with jitterbuffer simulator so every single file is properly synchronised. 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.

From customer: "I just wanted to write to say that Voipmonitor has been a HUGE help to us in managing our VOIP service in the face of South Africa's poor quality Internet environment. Aggregating the summary data provides fantastic information showing quality trends for individual customers, classes of customers and overall. The "clean" PCAP files are a huge help for debugging. What a pleasure to be able to go back in time and pull out a call from days ago and look at it packet by packet. So thanks so much for this very useful program!"

Key features

* Predicts MOS-LQE score according to ITU-T G.107 E-model 
* Records WAV files from G.711, for other codecs contacts support@voipmonitor.org 
* Fast C++ SIP/RTP packet analyzer 
* Detailed delay/loss statistics stored to MySQL 
* Each call is saved as standalone pcap file 
* Commercial web gui with nice loss/latency distribution

 

FUENTE: VOIPINFO.ORG