I ran xmllint over it and compared it to the results of 'sipp -sd uac'. It looks like the </recv> of the 100 Trying is copied twice, and you're missing the "<?xml version="1.0" encoding="ISO-8859-1" ?><!DOCTYPE scenario SYSTEM "sipp.dtd">" at the start.
Rob On 11 April 2014 18:28, Munzer,David J <mund...@ufl.edu> wrote: > Dear Rob, > > I made your recommended changes and checked for any missing </send> > requests and I am still receiving the "$ ./sipp -sf custom-uac.xml > 2014-04-12 02:16:43:685 1397283403.685916: Unable to load or > parse 'custom-uac.xml' xml scenario file." error. The local ip port is > left as [local ip port] since i am unsure which exact port is being > used. Please let me know if you find any more errors in the XML which > can be found below. > > Thankfully, > David > > <!-- 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]@10.0.0.160:5060 SIP/2.0 > Via: SIP/2.0/[transport] 10.0.0.210:[local_port];branch=[branch] > From: sipp > <sip:sipp@[10.0.0.210]:[local_port]>;tag=[pid]SIPpTag00[call_number] > To: sut <sip:[service]@10.0.0.160:5060> > Call-ID: [call_id] > CSeq: 1 INVITE > Contact: sip:sipp@10.0.0.210:[local_port] > Max-Forwards: 70 > Subject: Performance Test > Max-Forwards: 70 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: [len] > > v=0 > o=user1 53655765 2353687637 IN IP[local_ip_type] 10.0.0.210 > s=- > c=IN IP[media_ip_type] 10.0.0.160 > t=0 0 > m=audio [media_port] RTP/AVP 0 > a=rtpmap:0 PCMU/8000 > > ]]> > </send> > > <recv response="100" > optional="true"> > </recv> > 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]@10.0.0.160:5060 SIP/2.0 > Via: SIP/2.0/[transport] 10.0.0.210:[local_port];branch=[branch] > From: sipp > <sip:sipp@10.0.0.210:[local_port]>;tag=[pid]SIPpTag00[call_number] > To: sut <sip:[service]@10.0.0.160:5060>[peer_tag_param] > Call-ID: [call_id] > CSeq: 1 ACK > Contact: sip:sipp@[10.0.0.210]:[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]@10.0.0.160:5060 SIP/2.0 > Via: SIP/2.0/[transport] 10.0.0.210:[local_port];branch=[branch] > BYE sip:[service]@10.0.0.160:5060 SIP/2.0 > Via: SIP/2.0/[transport] 10.0.0.210:[local_port];branch=[branch] > From: sipp > <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number] > To: sut > <sip:[service]@10.0.0.160:5060>[peer_tag_param] > Call-ID: [call_id] > CSeq: 2 BYE > Contact: sip:sipp@10.0.0.210:[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> > > ------------------------------------------------------------------------------ > Put Bad Developers to Shame > Dominate Development with Jenkins Continuous Integration > Continuously Automate Build, Test & Deployment > Start a new project now. Try Jenkins in the cloud. > http://p.sf.net/sfu/13600_Cloudbees > _______________________________________________ > Sipp-users mailing list > Sipp-users@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/sipp-users ------------------------------------------------------------------------------ Put Bad Developers to Shame Dominate Development with Jenkins Continuous Integration Continuously Automate Build, Test & Deployment Start a new project now. Try Jenkins in the cloud. http://p.sf.net/sfu/13600_Cloudbees _______________________________________________ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
