[asterisk-users] voipmonitor.org

[Martin Vít]

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