If anyone want to replicate my test:
1. Call generator.
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:
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:
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.
sipp -sn uas -p 5070
where 5070 is the port of this answerer
I use the Yate 2.2 with G.729 codecs.
The routing is done via regex with this row:
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.