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) -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users