[Asterisk-Users] SIPP load testing - unexpected message - anyone using sipp sucessfully ?

[joachim]

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 at 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 at 192.168.0.101:5060>;tag=1
>To: sut <sip:service at 193.77.158.104:5060>;tag=as3e7533a6
>Call-ID: 1.3972.192.168.0.101 at sipp.call.id
>CSeq: 1 INVITE
>User-Agent: Asterisk PBX
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>Contact: <sip:service at 193.77.158.104>
>Content-Length: 0
>
>' .
>2005-02-08 00:23:36: Unexpected message for Call-ID
>'2.3972.192.168.0.101 at 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 at 192.168.0.101:5060>;tag=2
>To: sut <sip:service at 193.77.158.104:5060>;tag=as43cce205
>Call-ID: 2.3972.192.168.0.101 at sipp.call.id
>CSeq: 1 INVITE
>User-Agent: Asterisk PBX
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>Contact: <sip:service at 193.77.158.104>
>Content-Length: 0
>
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