Re: [Sipp-users] Loadtesting Asterisk with Sipp through Kamailio
That sounds logical. Thank you! I'll try the setup. -Original Message- From: Andrew Miller [mailto:andrew.mil...@crocodile-rcs.com] Sent: maandag 10 september 2012 14:44 To: sipp-users@lists.sourceforge.net Subject: Re: [Sipp-users] Loadtesting Asterisk with Sipp through Kamailio Grant, You could insert an extra Kamailio instance between SIPp and your test system. This would record-route and stay in dialog, allowing your real Kamailio to drop out of the dialog. Effectively your load generator consists of SIPp + a Kamailio (or any other) proxy Andy Miller Crocodile RCS Ltd On 08/09/2012 23:04, Michael Hirschbichler wrote: 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 Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- 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 -- Andrew Miller Crocodile RCS Ltd -- 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 -- 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
Re: [Sipp-users] Loadtesting Asterisk with Sipp through Kamailio
Grant, You could insert an extra Kamailio instance between SIPp and your test system. This would record-route and stay in dialog, allowing your real Kamailio to drop out of the dialog. Effectively your load generator consists of SIPp + a Kamailio (or any other) proxy Andy Miller Crocodile RCS Ltd On 08/09/2012 23:04, Michael Hirschbichler wrote: 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 Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- 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 -- Andrew Miller Crocodile RCS Ltd -- 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
Re: [Sipp-users] Loadtesting Asterisk with Sipp through Kamailio
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 Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- 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
[Sipp-users] Loadtesting Asterisk with Sipp through Kamailio
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
