Hello, I would like to introduce you VoIPmonitor.org solution
VoIPmonitor is open source network packet sniffer for SIP and RTP VoIP protocol for linux. VoIPmonitor was designed to analyze quality of VoIP calls based on network parameters - delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP protocol or SIP/RTP/RTCP protocols. VoIPmonitor can also decode speech and play it over the commercial WEB GUI or save it to disk as WAV. Supported codecs are G.711 alaw/ulaw and commercial plugins supports G.729a/G.723/iLBC/Speex/GSM. VoIPmonitor uses jitterbuffer simulator to keep both direction of call synchronized. You can download sniffer and trial WEB GUI from http://www.voipmonitor.org/download Datasheet http://www.voipmonitor.org/downloads/VoIPmonitor-GUI-datasheet.pdf We also offer hardware solution and custom development. VoIPmonitor.org Team
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
