Sipp - SIP Performance Tester

Sipp is a performance tester for the SIP protocol. It comes with a few basic SipStone user-agents scenarios (UAC & UAS), establishing and releasing multiple calls with the INVITE and BYE methods.

  • Asterisk configuration for SIP (non-rtp listening) test
Create a context like so:

context testing {
1 => {
//modify for your setup.. make it play something, join a queue, whatever. I found that joining a queue works best, queue_helpdesk is my helpdesk macro :)

Configure a sip "friend" to the IP address of your testing box, obviously substituting ulaw with alaw or another codec (I run two asterisk boxes, so used my second one)


and then run from the following command

sipp -sn uac -s 1 -d 100000 -l 256

This pretty much boils down to:
dial through to sip:[email protected] (in the context testing), with a pause of 100000ms before hanging up, and a total concurrent call limit of 256.
I still have to work out how to do RTP testing. Will update with more details :)

It can also read XML scenario files describing any performance testing configuration for SIP.

SIPP can run as UAS also so I setup a UAC and UAS to receive the calls,
the problem I saw was that when the SIPP UAC sends a BYE to * it hangups
the channel but doesn�t send the corresponding BYE to the SIPP UAS so
the UASS think that the call wasn�t teared down.
So I noted if I used nat=yes (in Asterisk config sip.conf) for SIPP UAS and UAC everything went fine
I think maybe this is related to the sockets used for each SIPP thread
since by default it uses the same socket for each call.

See also

Created by: flavour, Last modification: Thu 16 of Aug, 2007 (23:11 UTC) by fujin
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