Hi Adi,
in sipp, a "call" is a single pass through the from start to end,
regardless whether it is an actual call-controlling (INVITE-200-BYE-200)
dialog, a REGISTER-401-REGISTER-200 sequence, or e.g. a
SUBSCRIBE-200-NOTIFY-200-... dialog.
So in your register_client.xml, the "call" ends by receiving the 200
response to the Register with authentication header.
The number of balls in the air is determined by total flight time of a
single ball (in your case, Σ_t = twice the network round-trip time plus
Kamailio's processing time of non-authenticated and of authenticated
REGISTER) and the number of balls you manage to throw until the first
one falls back. If the Kamailio is quick enough and the network path is
short enough, you may actually get just tens of really concurrent calls
for 100 calls in 80 ms. So I'd recommend to track the time of the first
REGISTER and of the 200 response to the 2^nd one (either using the sipp
logging facility or using tcpdump/Wireshark) and modify -r and -rp
depending on the result (like -r 1000 -rp _Σ_t ).
Also think about the time it takes to send a message through a network
interface of a given bitrate: a 1250-byte message represents about 10000
bits to be sent, so on a 100 Mbit/s Ethernet, you cannot send more than
10 such messages per millisecond, i.e. 800 in 80 milliseconds, if there
is no other traffic on the interface.
If you say that -r X -rp Y -l Z -m T doesn't do anything, try to add the
parameters one-by-one, left-to-right, to find out which one breaks it,
and then start omitting them one by one (-r -rp, -r -rp -l, -r -rp -l
-m, -r -l -m, ...) to see if some combinations are not a problem, rather
than individual parameters.
This sentence is totally cryptic to me, it seems you were in a hurry:
Also is this test to "create 100 calls during each 80 milliseconds up
to a limit of 1000
calls in progress" means that the calls, here register method calls
not ths calls calls,
are keep-alive simultaneous till the 2000 one ends?
If what I wrote is not the answer, please translate the sentence.
P.
Dne 21.3.2016 v 16:00 adrian Albu napsal(a):
Díky Sindelka,
./sipp -sf register_client.xml -inf user21.csv -i local_ip
kamailio_ip:k_port -trace_err -r 100
is kind of working using
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="register_client">
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0]@[field1]>;tag=[call_number]
To: <sip:[field0]@[field1]>
Call-ID: [call_id]
CSeq: 1 REGISTER
Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 5
Expires: 1800
User-Agent: SIPp/Linux
Content-Length: 0
]]>
</send>
<recv response="401" auth="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0]@[field1]>;tag=[call_number]
To: <sip:[field0]@[field1]>
Call-ID: [call_id]
CSeq: 2 REGISTER
Contact: sip:[field0]@[local_ip]:[local_port]
[field2]
Max-Forwards: 5
Expires: 1800
User-Agent: SIPp/Linux
Content-Length: 0
]]>
</send>
<recv response="200">
</recv>
</scenario>
here is the result:
------------------------------ Scenario Screen -------- [1-9]: Change
Screen --
Call-rate(length) Port Total-time Total-calls Remote-host
100.0(0 ms)/1.000s 5060 20.05 s 2005
kamailio_ip:k_port(UDP)
2 new calls during 0.021 s period 1 ms scheduler resolution
1 calls (limit 300) Peak was 1 calls, after 0 s
0 Running, 2008 Paused, 2 Woken up
0 dead call msg (discarded) 0 out-of-call msg (discarded)
3 open sockets
Messages Retrans Timeout
Unexpected-Msg
REGISTER ----------> 2005 0 0
401 <---------- 2005 0 0 0
REGISTER ----------> 2005 0 0
200 <---------- 2004 0 0 0
------------------------------ Test Terminated
--------------------------------
----------------------------- Statistics Screen ------- [1-9]: Change
Screen --
Start Time | 2016-03-21 15:50:31.598871 1458571831.598871
Last Reset Time | 2016-03-21 15:50:51.632419 1458571851.632419
Current Time | 2016-03-21 15:50:51.654458 1458571851.654458
-------------------------+---------------------------+--------------------------
Counter Name | Periodic value | Cumulative value
-------------------------+---------------------------+--------------------------
Elapsed Time | 00:00:00:022000 | 00:00:20:055000
Call Rate | 90.909 cps | 99.975 cps
-------------------------+---------------------------+--------------------------
Incoming call created | 0 | 0
OutGoing call created | 2 | 2005
Total Call created | | 2005
Current Call | 1 |
-------------------------+---------------------------+--------------------------
*Successful call | 2 | 2004*
Failed call | 0 | 0
-------------------------+---------------------------+--------------------------
Call Length | 00:00:00:003000 | 00:00:00:003000
------------------------------ Test Terminated
--------------------------------
But if I want to put more params to stress it a bit, like
./sipp -sf register_client.xml -inf user21.csv -i local_ip
kamailio_ip:k_port -trace_err -r 100 -rp 80 -l 1000 -m 2000
is doing nothing.
Also is this test to "create 100 calls during each 80 milliseconds up
to a limit of 1000
calls in progress" means that the calls, here register method calls
not ths calls calls,
are keep-alive simultaneous till the 2000 one ends?
I guess the ones from -r flag, here -r 100 , would be concurrent, but
what is the default for the time till the sip session expires?
I want to be sure I have an minimum number of sessions for a while to
see it works. Maybe your explanation is clear but all these is new to me,
I want to learn it but maybe am a bit impatient :)
Thanks again,
Adrian
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On Monday, 21 March 2016, 10:38, sindelka <sinde...@ttc.cz> wrote:
Bună ziua Adi,
look at
http://sipp.sourceforge.net/doc/reference.html#Online+help+%28-h%29
with special care about the meaning of -r, -rp, -l and -m command line
parameters.
Example:
sipp -sf your_scenario_name.xml -r 100 -rp 80 -l 1000 -m 2000
will create 100 calls during each 80 milliseconds up to a limit of
1000 calls in progress. Terminated calls will be replaced by new ones
until the total number of calls will reach 2000. After that, no new
calls will be added when the running ones finish. The scenario
execution ends when the last call finishes.
This is unrelated to the contents of the injection file and/or to
whether it exists at all.
The limitations of your hardware may cause the actual performance to
be lower, especially if you are going to send RTP as part of the load
test.
P.
Dne 21.3.2016 v 9:47 adrian Albu napsal(a):
Hi,
I was searching online for a way to test simultaneous sip sessions,
for example 1000 connected on it at the same time.
could not find an example,
the only starting point I found was:
http://comments.gmane.org/gmane.comp.telephony.sipp.user/3934
Is there an example somewhere?
Thanks,
Adrian
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