Attaching Scenario run as it is, may be spacelines, CRLF get changed while copy/paste.
Best Regards,Sakharam Thorat. From: [email protected] To: [email protected] Date: Tue, 23 Sep 2014 15:57:03 +0530 CC: [email protected] Subject: Re: [Sipp-users] SIP-I Hi i tried to run following scenario on sipp 3.4.1, saw outgoing isup msg in wireshark as part of sip nsg. <?xml version="1.0" encoding="ISO-8859-1" ?><!DOCTYPE scenario SYSTEM "sipp.dtd"> <!-- This program is free software; you can redistribute it and/or --><!-- modify it under the terms of the GNU General Public License as --><!-- published by the Free Software Foundation; either version 2 of the --><!-- License, or (at your option) any later version. --><!-- --><!-- This program is distributed in the hope that it will be useful, --><!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --><!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --><!-- GNU General Public License for more details. --><!-- --><!-- You should have received a copy of the GNU General Public License --><!-- along with this program; if not, write to the --><!-- Free Software Foundation, Inc., --><!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --><!-- --><!-- Sipp default 'uac' scenario. --><!-- --> <scenario name="Basic Sipstone UAC"> <!-- In client mode (sipp placing calls), the Call-ID MUST be --> <!-- generated by sipp. To do so, use [call_id] keyword. --> <send retrans="500"> <![CDATA[ INVITE sip:[email protected] SIP/2.0 From: <sip:[email protected]>;tag=e27b3 To: <sip:[email protected]> Call-Id: scb2f9c3f297d0197b29f4591169fcc23 Cseq: 25077 INVITE Session-Expires: 1800 Min-Expires: 90 Content-Type:multipart/mixed;boundary=level3-boundary Content-Length: 675 Expires: 180 Date: Tue, 01 Mar 2011 04:19:12 GMT Max-Forwards: 69 User-Agent: wxCommunicator Accept-Language: en Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,INFO,PRACK,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REGISTER,REFER Supported: replaces,timer,100rel,from-change,norefersub Via: SIP/2.0/UDP 173.170.11.84:5060;branch=z9hG4bK-83c6ce099e7d;rport Contact: <sip:[email protected]:5060> Content-Length: [len] MIME-Version: 1.0 --level3-boundary Content-Type: application/sdp v=0 o=sipX 5 10 IN IP4 192.168.1.138 s=call c=IN IP4 192.168.1.138 t=0 0 m=audio 9000 RTP/AVP 96 97 a=candidate:0 t UDP 1.0 192.168.1.138 9000 a=candidate:0 t UDP 1.0 192.168.1.138 9001 a=candidate:1 t UDP 0.5 5.210.195.106 9000 a=candidate:1 t UDP 0.5 5.210.195.106 9001 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:97 speex/32000/1 a=fmtp:97 mode=4 a=ptime:20 --level3-boundary Content-Type: application/isup;base=itu-t92+;version=itu Content-Disposition: session;handling=optional \x01\x00\x20\x00\x00\x03\x02\x06\x04\x01\x10\x21\x43\x0a\x08\x01\x15\x44\x21\x43\x65\x87\x09\x00 --level3-boundary-- ]]> </send> <recv response="100" optional="true"> </recv> <recv response="180" optional="true"> </recv> <recv response="183" optional="true"> </recv> <!-- By adding rrs="true" (Record Route Sets), the route sets --> <!-- are saved and used for following messages sent. Useful to test --> <!-- against stateful SIP proxies/B2BUAs. --> <recv response="200" rtd="true"> </recv> <!-- Packet lost can be simulated in any send/recv message by --> <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --> <send> <![CDATA[ ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <!-- This delay can be customized by the -d command-line option --> <!-- or by adding a 'milliseconds = "value"' option here. --> <pause/> <!-- The 'crlf' option inserts a blank line in the statistics report. --> <send retrans="500"> <![CDATA[ BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 2 BYE Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <recv response="200" crlf="true"> </recv> <!-- definition of the response time repartition table (unit is ms) --> <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/> <!-- definition of the call length repartition table (unit is ms) --> <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> </scenario> To copile Sipp 3.4.1 , can take look at http://techvick.blogspot.in/2014/09/how-to-intsall-sipp-on-ubuntu.html Best Regards,Sakharam Thorat. From: [email protected] To: [email protected]; [email protected] Subject: RE: [Sipp-users] SIP-I Date: Tue, 23 Sep 2014 15:09:55 +0530 Hello Sakharam/Fabio, I am not sure if this code merge has really worked with real soft switch or PSTN GW testing. check this :: http://sourceforge.net/p/sipp/patches/34/ this patch seem to be tested back to back.. @fabio : is it some Network deployment you want to test ? Thanks -Abinash From: [email protected] To: [email protected] Date: Tue, 23 Sep 2014 13:32:21 +0530 CC: [email protected] Subject: Re: [Sipp-users] SIP-I Hi, Please take a look at following scenario, <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"><!-- This program is free software; you can redistribute it and/or --> <!-- modify it under the terms of the GNU General Public License as --> <!-- published by the Free Software Foundation; either version 2 of the --> <!-- License, or (at your option) any later version. --> <!-- --> <!-- This program is distributed in the hope that it will be useful, --> <!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --> <!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --> <!-- GNU General Public License for more details. --> <!-- --> <!-- You should have received a copy of the GNU General Public License --> <!-- along with this program; if not, write to the --> <!-- Free Software Foundation, Inc., --> <!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --> <!-- --> <!-- Sipp default 'uac' scenario. --> <!-- --><scenario name="Basic Sipstone UAC"> <!-- In client mode (sipp placing calls), the Call-ID MUST be --> <!-- generated by sipp. To do so, use [call_id] keyword. --> <send retrans="500"> <![CDATA[INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]> Call-ID: [call_id] CSeq: 1 INVITE Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Type: multipart/mixed;boundary=level3-boundary Content-Length: [len] MIME-Version: 1.0--level3-boundary Content-Type: application/sdpv=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 --level3-boundary Content-Type: application/isup;base=itu-t92+;version=itu Content-Disposition: session;handling=optional\x01\x00\x20\x00\x00\x03\x02\x06\x04\x01\x10\x21\x43\x0a\x08\x01\x15\x44\x21\x43\x65\x87\x09\x00 --level3-boundary--]]> </send><recv response="100" optional="true"> </recv><recv response="180" optional="true"> </recv><recv response="183" optional="true"> </recv><!-- By adding rrs="true" (Record Route Sets), the route sets --> <!-- are saved and used for following messages sent. Useful to test --> <!-- against stateful SIP proxies/B2BUAs. --> <recv response="200" rtd="true"> </recv><!-- Packet lost can be simulated in any send/recv message by --> <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --> <send> <![CDATA[ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0]]> </send><!-- This delay can be customized by the -d command-line option --> <!-- or by adding a 'milliseconds = "value"' option here. --> <pause/><!-- The 'crlf' option inserts a blank line in the statistics report. --> <send retrans="500"> <![CDATA[BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 2 BYE Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Type: application/isup;base=itu-t92+;version=itu Content-Disposition: session;handling=optional Content-Length: [len]\x0c\x02\x00\x02\x87\x90]]> </send><recv response="200" crlf="true"> </recv><!-- definition of the response time repartition table (unit is ms) --> <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/><!-- definition of the call length repartition table (unit is ms) --> <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/></scenario> Best Regards,Sakharam Thorat. Date: Tue, 23 Sep 2014 00:55:54 -0700 From: [email protected] Subject: SIP-I To: [email protected] Hi, do you know how to embed ISUP packets inside SIP messages ? I read someone made a patch. Is it mainstream ? Do you have a small example ? thank you Fabio ------------------------------------------------------------------------------ Meet PCI DSS 3.0 Compliance Requirements with EventLog Analyzer Achieve PCI DSS 3.0 Compliant Status with Out-of-the-box PCI DSS Reports Are you Audit-Ready for PCI DSS 3.0 Compliance? Download White paper Comply to PCI DSS 3.0 Requirement 10 and 11.5 with EventLog Analyzer http://pubads.g.doubleclick.net/gampad/clk?id=154622311&iu=/4140/ostg.clktrk _______________________________________________ Sipp-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/sipp-users ------------------------------------------------------------------------------ Meet PCI DSS 3.0 Compliance Requirements with EventLog Analyzer Achieve PCI DSS 3.0 Compliant Status with Out-of-the-box PCI DSS Reports Are you Audit-Ready for PCI DSS 3.0 Compliance? Download White paper Comply to PCI DSS 3.0 Requirement 10 and 11.5 with EventLog Analyzer http://pubads.g.doubleclick.net/gampad/clk?id=154622311&iu=/4140/ostg.clktrk _______________________________________________ Sipp-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/sipp-users
sipp-uac.xml
Description: XML document
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