Hi Yasir,
half of your questions is actually irrelevant to SIPp :-) In the
Asterisk configuration, you've stated to continue by responding the
incoming INVITE with Ringing if the routing step
exten => 8001,1,Dial(SIP/8001,20)
returns with any result. So I suspect that you haven't registered any
SIP phone as extension 8001, and maybe you haven't even defined it in
sip.conf, so the Asterisk doesn't know where to send the INVITE
specified in step 1, so it returns immediately to step 2.
Back to the SIPp setup necessary for your application: if you want to
stress-test any SIP PBX using SIPp, you need two SIPp instances: one
running the UAC scenario, i.e. acting as the calling side, where you
"modulate" the call rate and number of concurrent calls, and another one
running an UAS scenario, i.e. acting as called party, which responds all
calls with no delays (with 4xx or with 200 followed by sent or expected
BYE depending on what you want to test). So at the called side, the
embedded UAS scenario may not be enough.
If you want to test calls to a registered local SIP user on Asterisk in
particular (because processing of such calls takes a specific path
through Asterisk's code),which seems to be your case given how you've
described it, the SIPp scenario imitating the called party must include
registration which is not the simplest task - to do that, you have to
create a pair of scenarios at the called side, an UAS one which expects
an incoming INVITE or a Cmd from the other scenario. The received INVITE
is processed normally, the received Cmd triggers sending of a REGISTER.
I've written more on this topic here at sipp-users less than a month ago.
Also, the -ap is necessary to authorize the calling or registering
party, i.e. it must correspond to the From uri of the request sent. So
for calls *to* 8001, -ap for 8001 is useless. Plus Asterisk normally
rejects calls which come in with a locally defined number in From via a
trunk.
As for rejection of calls coming from unknown calling parties, that's
purely an Asterisk question I don't feel competent to answer.
Just a final remark, some time ago there was a problem that Asterisk had
first allocated all resources needed to process a call and only then it
checked whether the call was allowed, which was causing it it to hang
under floods of INVITEs. So if you manage to generate a high enough rate
of call attempts, you may have to restart the Asterisk after such test.
But it may need a farm of machines running SIPp, or not be possible at
all if the Asterisk code has changed since.
Pavel
Dne 25.7.2016 v 5:49 yasir al-ibrahem napsal(a):
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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