Hello dears,
I've been using SIPp for a week now, and I need some help to understand the
expected message flow please.
My test procedure was:
(1) I've configured an extension on my Asterisk as '8001/8001mike' in
sip.conf
(2) I configured a simple dialplan for that extension so that it can be
dialled, as:
exten => 8001,1,Dial(SIP/8001,20)
exten => 8001,2,Ringing()
exten => 8001,3,Answer()
exten => 8001,4,Hangup()
(3) on another machine, I installed SIPp and ran it as below
# sipp -sn uac <Asterisk IP> -s 8001 -ap 8001mike -r 1 -l 5 -m 5
Results:
(1) calls were all successful
(2) The call flow on Asterisk was
--> INVITE
<-- 100 Trying
<-- 200 OK
--> ACK
--> BYE
<-- 200 OK
Questions:
(1) Asterisk generated the above messages without issuing any actual
messages towards the 8001 as a called party. This isn't a reliable stress
test for Asterisk since it didn't interact with 8001.
Is that a correct test procedure ?
Is there a way to make SIPp traffic tests with Asterisk to generate SIP
messages completely to both parties ?
note: I know that If I start the sipphone (e.g. X-Lite, Zoiper) that I get
the ringing and I must answer it manually, but I can't answer calls
manually at a rate of 50 calls per second.
(2) This is for Asterisk.
SIPp generates INVITE messages with a caller as 'sipp@<IP>'. Do you know
how to force Asterisk to reject INVITE messages if the caller isn't a
registered client ?
T
hanks a lot
Best Regards,
*Yasir Saad Al-Ibrahem+1-312-428-0301*
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