[Salvatore Frandina <salvatore.frandina@xxxxxxxxx>]

Hi,

I'm using Wireshark to calculate the bandwidth of SIP based audioconference. I use Wireshark 1.2.6 rev 31702 on Windows vista 32 bit sp2 with bria 2.4 softphone. 
I have a question about the RTP statistics. Telephony->RTP->Show all streams->analyze->save as CSV.

I have calculate the bandwidth histogram with MATLAB using the seventh column of CSV file.

If in the conference all members use the same audio codec, I get correct value as reported here http://www.voip-info.org/wiki/view/Bandwidth+consumption.

If in the conference the members use different audio codec (like GSM, G.729 and G.711a/u) I get a mean bandwidth value about 80 kb/s due to G.711a/u codec and a values (small percentage) up to 120 kb/s.

Is it normal? Where could be the problem?

Thank

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Salvatore Frandina
website: http://frandinas.altervista.org

mail: salvatore.frandina@xxxxxxxxx

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