On 8.5.2010 00:40, Jeff Brower wrote: > 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?
I've not tested limits but capturing 15 voip calls takes 3-4% on Core2 2.40GHz. Complexity in worst case is O(N^2) where N is number of calls. Packets are matched as llinear list of IP and port. If this will be limit, it could be rewriten to hash table O(N) > How does it do MOS prediction if low bitrate codecs are being used > (G729, AMR, etc)? > It is calibrated only to G.711 with PLC for now but I'm planing adding equations for G.729 and iLBC. MV -- _____________________________________________________________________ -- 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