Here is mine example of sipp to send calls RTP media:
You can get different RTP files : https://mpbx.a4business.com/pcap.tgz ,
which I’ve found and converted from wav using wav2rtp cli app.
( concatenate files to get longer duration )
This part of scenario makes the streaming of RTP :
To start just one call for test, with certain CallerID and destination
number, I used info files, and run it with command:
sipp -sf ./call.xml 192.168.1.11 -inf ./cli/986533.csv -s 123456 -l 1 -m 1
-d 5000 -trace_logs -trace_msg -default_behaviors all -recv_timeout 5000
-cp 15762 -mp 34390 -aa
where 192.168.1.11 - remote asterisk server.
-s 123456 - CallerID number
-recv_timeout - SIPP send call, and giving up if no answer during
first 5 sec
-d : max call duration of connected call./
The file cli/986533.csv has only two lines, and used in scenario as field0
( thats the only way to put Source Caller ID which I’ve found for sipp )
it was required to get a real duration of the call once it completed ( if
not connected - return 0 ), for billing. so this just a part of all script
which I’ve made for it.
The “bottle-neck” was how many connection per second asterisk is able to
i,e. if I generate longer duration calls ( 5 minutes ), 10 new call per
second - it could grow up to max 3000 concurrent calls on 1GB NIC card , it
made almost 250MBit/s of one way traffic (from sipp to asterisk ) .
if generate 20 calls per second, with 2m duration, asterisk dies (core
dump) after ~1000 concurrent calls ( while max should be 2400 with this