Hello,
I'm trying to perform a loadtest on our Asterisk machines using Sipp, but there
is a SIP Proxy(Kamailio) in between. Kamailio acts as a loadbalancer for our
Asterisk machines, so it only remains in the dialog during the initial INVITE,
TRYING and 200 OK. The ACK should be sent by SIPp directly to the Asterisk
machines. This is where I'm having problems. I can't get SIPp to send the ACK
directly to the Asterisk machines, without enabling record-route. When I enable
record-route Kamailio stays in between, but this is not a representative
loadtest for our live platform, since record-route is disabled on live. The
contact header is set properly, but SIPp refuses to send the ACK directly.
Is this a limitation of SIPp when used in this kind of setup, or am I doing
something wrong? I have already loadtested Asterisk without Kamailio in
between, that went fine.
Regards,
Grant
------------------------------------------------------------------------------
Live Security Virtual Conference
Exclusive live event will cover all the ways today's security and
threat landscape has changed and how IT managers can respond. Discussions
will include endpoint security, mobile security and the latest in malware
threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/
_______________________________________________
Sipp-users mailing list
Sipp-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/sipp-users
