From: Michael H. <si...@hi...> - 2012-09-08 22:20:06
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Hi, this won't work AFAIK - sipp sends requests only to one remote socket during a call. br Michael Am 07.09.2012 08:06, schrieb Grant Bagdasarian: > 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 > Sip...@li... > https://lists.sourceforge.net/lists/listinfo/sipp-users > |