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
'[EMAIL PROTECTED]': 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:[EMAIL PROTECTED]:5060>;tag=1
To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as3e7533a6
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
' .
2005-02-08 00:23:36: Unexpected message for Call-ID
'[EMAIL PROTECTED]': 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:[EMAIL PROTECTED]:5060>;tag=2
To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as43cce205
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
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