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) -- _____________________________________________________________________ -- 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
