asterisk users - Feb 2005 - SIPP load testing - unexpected message - anyone using sipp sucessfully ?

Hi,

I'd like to test Asterisk performance under more concurrent sip calls. I use
Sipp, but do get "Unexpected message for Call-ID ...", so I wonder if
anyone
is using sipp succesfully with Asterisk and is willing to share more info
about his solution ...

Any other convenient way to load test Asterisk ?  Is sipp the right tool ?

Thanks in advance,

regards,

Rob.



sipp: The following events occured:
2005-02-08 00:23:36: Unexpected message for Call-ID
'1.3972.192.168.0.101@sipp.call.id': while expecting '100'
response,
received 'SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.101:5060;received=193.77.90.224;rport=5060
From: sipp <sip:sipp@192.168.0.101:5060>;tag=1
To: sut <sip:service@193.77.158.104:5060>;tag=as3e7533a6
Call-ID: 1.3972.192.168.0.101@sipp.call.id
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:service@193.77.158.104>
Content-Length: 0

' .
2005-02-08 00:23:36: Unexpected message for Call-ID
'2.3972.192.168.0.101@sipp.call.id': while expecting '100'
response,
received 'SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.101:5060;received=193.77.90.224;rport=5060
From: sipp <sip:sipp@192.168.0.101:5060>;tag=2
To: sut <sip:service@193.77.158.104:5060>;tag=as43cce205
Call-ID: 2.3972.192.168.0.101@sipp.call.id
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:service@193.77.158.104>
Content-Length: 0

SIP will get you no RTP, meaning it only works with SIP headers.
Asterisks CPU usage is mainly coming from RTP handling.

We glued something together that will work for RTP too, you can download
it from:
http://www.astertest.com/forum/viewtopic.php?t=4

As the moment it only seems to work for non authenticated SIP calls, but
it does support RTP.

Other options are commercial tools such as WINSIP etc. (more call
generators + descriptions can be found in the ppt presentation on
www.astertest.com)


SIPP works for asterisk testing too, but you need the correct
commandline. What did you use  ?

Joachim





Robert Rozman wrote:
>Hi,
>
>I'd like to test Asterisk performance under more concurrent sip calls. I
use
>Sipp, but do get "Unexpected message for Call-ID ...", so I wonder
if anyone
>is using sipp succesfully with Asterisk and is willing to share more info
>about his solution ...
>
>Any other convenient way to load test Asterisk ?  Is sipp the right tool ?
>
>Thanks in advance,
>
>regards,
>
>Rob.
>
>
>
>sipp: The following events occured:
>2005-02-08 00:23:36: Unexpected message for Call-ID
>'1.3972.192.168.0.101@sipp.call.id': while expecting '100'
response,
>received 'SIP/2.0 404 Not Found
>Via: SIP/2.0/UDP 192.168.0.101:5060;received=193.77.90.224;rport=5060
>From: sipp <sip:sipp@192.168.0.101:5060>;tag=1
>To: sut <sip:service@193.77.158.104:5060>;tag=as3e7533a6
>Call-ID: 1.3972.192.168.0.101@sipp.call.id
>CSeq: 1 INVITE
>User-Agent: Asterisk PBX
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>Contact: <sip:service@193.77.158.104>
>Content-Length: 0
>
>' .
>2005-02-08 00:23:36: Unexpected message for Call-ID
>'2.3972.192.168.0.101@sipp.call.id': while expecting '100'
response,
>received 'SIP/2.0 404 Not Found
>Via: SIP/2.0/UDP 192.168.0.101:5060;received=193.77.90.224;rport=5060
>From: sipp <sip:sipp@192.168.0.101:5060>;tag=2
>To: sut <sip:service@193.77.158.104:5060>;tag=as43cce205
>Call-ID: 2.3972.192.168.0.101@sipp.call.id
>CSeq: 1 INVITE
>User-Agent: Asterisk PBX
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>Contact: <sip:service@193.77.158.104>
>Content-Length: 0
>
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>Asterisk-Users@lists.digium.com
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On Tue, 08 Feb 2005 10:16:46 +0200, joachim <zoachien@securax.org>
wrote:
> SIPP works for asterisk testing too, but you need the correct
> commandline. What did you use  ?
Perhaps you can just give us the _correct_ command line for those of
us who are unknowing? :)

Thanks,
Leif Madsen.
http://www.leifmadsen.com

Hi,

I used this command line :

sipp -sn uac URL_of_*_server -trace_err

Is there any better ?

Regards,

Rob.