<html>
<head>
<style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 10pt;
font-family:Verdana
}
--></style>
</head>
<body class='hmmessage'>
<BR><BR><BR><BR> <BR>
> Date: Mon, 10 May 2010 09:39:55 +0200<BR>> From: vit@lam.cz<BR>> To: asterisk-users@lists.digium.com<BR>> Subject: Re: [asterisk-users] voipmonitor.org<BR>> <BR>> On 8.5.2010 00:40, Jeff Brower wrote:<BR>> > Martin-<BR>> ><BR>> > <BR>> >> checkout new open source voipmonitor.org SIP packet sniffer. I've<BR>> >> developed it for my telco company and I've decided to share it.<BR>> >> Testing and contributions are welcome!<BR>> >><BR>> >> VoIPmonitor is open source live network packet sniffer which analyze<BR>> >> SIP and RTP protocol. It can run as daemon or analyzes already<BR>> >> captured pcap files. For each detected VoIP call voipmonitor<BR>> >> calculates statistics about loss, burstiness, latency and predicts MOS<BR>> >> (Meaning Opinion Score) according to ITU-T G.107 E-model. These<BR>> >> statistics are saved to MySQL database and each call is saved as pcap<BR>> >> dump. Web PHP application (it is not part of open source sniffer)<BR>> >> filters data from database and graphs latency and loss distribution.<BR>> >> Voipmonitor also detects improperly terminated calls when BYE or OK<BR>> >> was not seen. To accuratly transform latency to loss packets,<BR>> >> voipmonitor simulates fixed and adaptive jitterbuffer.<BR>> >> <BR>> > How many channels can it handle simultaneously? <BR>> <BR>> I've not tested limits but capturing 15 voip calls takes 3-4% on Core2<BR>> 2.40GHz. Complexity in worst case is O(N^2) where N is number of calls.<BR>> Packets are matched as llinear list of IP and port. If this will be<BR>> limit, it could be rewriten to hash table O(N)<BR>> <BR>> > How does it do MOS prediction if low bitrate codecs are being used<BR>> > (G729, AMR, etc)?<BR>> > <BR>> <BR>> It is calibrated only to G.711 with PLC for now but I'm planing adding<BR>> equations for G.729 and iLBC.<BR>> <BR>> MV<BR>> <BR><BR>
Maybe this question is out little but is the same context. I need read the VoIP packets and order all this packets in another place to get the audio. The idea is can record a call using directly the packets.<BR>
I know asterisk can record but my problem is that I have Avaya and asterisk working togheter and I can not record by Avaya and somebody tells me this idea to sniff the VoIP packets order after the call.<BR>
<BR>
I am seeing the code for VoIp monitor<BR>
Is it so stupid??<BR>
<BR>
<BR>
TIA<BR> <br /><hr /> <a href='' target='_new'></a></body>
</html>