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 To :  Diana Cionoiu <diana-liste@v...>
 From :  Alfred Stainer <alfred.stainer@g...>
 Subject :  Re: [yate] Core dump
 Date :  Wed, 28 Apr 2010 09:40:44 +0200
If anyone want to replicate my test:

1. Call generator.

To generate the calls I use sipp (sipp.sourceforge.net). The configuration
file I use is uac_pcap.xml: http://pastebin.com/sA2d9M11

In this xml file there is a command to play a "g729.pcap" that contains RTP
voice data. This file can be created with wireshark (or tcpdump) capturing a
call between two sip clients (I used 2 hw phone configured in g.729) and
filtering RTP data.

to generate the call I run:

   sipp -inf called.csv -r 5 -rp 1000 -l 500 -t un -sf uac_pcap.xml
192.168.1.1

where 192.168.1.1 is the IP of Yate and there are generated 5 calls per
second with a maximum of 500 concurrent calls.

The file called.csv is a txt file that contains a first row with 'RANDOM'
and others rows with calling;called like:

RANDOM
12345635413;133481209413;
12345666375;143481209768;
12345614661;153481204630;
12345681083;163481202653;
...

This file is used to generate random calling and called in sip INVITE. If
you want, you can have only 1 calling and 1 called with a two rows
called.csv file:
RANDOM
12345635413;133481209413;
or you can modify the xml file changing [field0] and [filed1] with your
calling/called and invoking the sipp without "-inf called.csv" parameter.

2. Call answerer.
I use sipp that can run in the same PC of call generator.
I run:
   sipp -sn uas -p 5070

where 5070 is the port of this answerer


3. Yate
I use the Yate 2.2 with G.729 codecs.
The routing is done via regex with this row:

.*=sip/sip:@1...:5070;timeout=145000

where 192.168.1.2 is the IP where are running call generator and call
answerer.

In my test, the two PC (call genarator+answerer and yate) are connected with
a dedicated 100Mb switch.

Regards,

Alfred



If anyone want to replicate my test:
 
1. Call generator.
 
To generate the calls I use sipp (sipp.sourceforge.net). The configuration file I use is uac_pcap.xml: http://pastebin.com/sA2d9M11 
 
In this xml file there is a command to play a "g729.pcap" that contains RTP voice data. This file can be created with wireshark (or tcpdump) capturing a call between two sip clients (I used 2 hw phone configured in g.729) and filtering RTP data.
 
to generate the call I run:
 
   sipp -inf called.csv -r 5 -rp 1000 -l 500 -t un -sf uac_pcap.xml 192.168.1.1
 
where 192.168.1.1 is the IP of Yate and there are generated 5 calls per second with a maximum of 500 concurrent calls.
 
The file called.csv is a txt file that contains a first row with 'RANDOM' and others rows with calling;called like:
 
RANDOM
12345635413;133481209413;
12345666375;143481209768;
12345614661;153481204630;
12345681083;163481202653;
... 
 
This file is used to generate random calling and called in sip INVITE. If you want, you can have only 1 calling and 1 called with a two rows called.csv file:
RANDOM
12345635413;133481209413;
or you can modify the xml file changing [field0] and [filed1] with your calling/called and invoking the sipp without "-inf called.csv" parameter.
 
2. Call answerer.
I use sipp that can run in the same PC of call generator.
I run:
   sipp -sn uas -p 5070
 
where 5070 is the port of this answerer
 
 
3. Yate
I use the Yate 2.2 with G.729 codecs.
The routing is done via regex with this row:
 
.*=sip/sip:@1...:5070;timeout=145000
 
where 192.168.1.2 is the IP where are running call generator and call answerer.
 
In my test, the two PC (call genarator+answerer and yate) are connected with a dedicated 100Mb switch.
 
Regards,
 
Alfred