[asterisk-users] voipmonitor.org

[ram]

On Mon, May 10, 2010 at 1:09 PM, Martin Vít <vit at lam.cz> wrote:

> 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.
>
when are you expecting to release

Ram
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