Thank you - I'll take a look at this this evening. In the meantime, can you run SIPp with the additional command-line options "-trace_err -trace_calldebug -trace_msg"?
This will produce three log files, one ending _messages.log, one ending _errors.log and one ending with _calldebug.log - it would be useful if you could send me those. Best, Rob On 7 January 2013 15:09, Rajesh Bansal <bansal.rajes...@gmail.com> wrote: > Hi Rob, > > I am using sipp-3.3 at RHEL 5.3 (32 bit). > > I want to test following SIP scenario with SIPP. This is produced by making > a successful SIP call from Xlite softphone client to freeswitch (Application > Hosted). > > CLIENT Freeswtich > > Invite ------------> > <---------- 100 Trying > <---------- 183 with SDP > <-- Media Play --> > Bye ------------> > <--------------- 200 OK > > > I am getting following > > Invite ------------> > <----------- 100 Trying > <----------- 183 with SDP > Cancel ------------> > <----------- 200 OK > > Following is my scenario file. > > > <?xml version="1.0" encoding="ISO-8859-1" ?> > <!DOCTYPE scenario SYSTEM "sipp.dtd"> > > <scenario name="UAC with media"> > <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: [field0] > <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number] > To: sut <sip:[service]@[remote_ip]:[remote_port]> > Call-ID: [call_id] > CSeq: 1 INVITE > Contact: sip:field0@[local_ip]:[local_port] > Max-Forwards: 70 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: [len] > > v=0 > o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] > s=- > c=IN IP[local_ip_type] [local_ip] > t=0 0 > m=audio [auto_media_port] RTP/AVP 8 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11,16 > > ]]> > > </send> > > <recv response="100" optional="true"> > </recv> > > <recv response="183"> > </recv> > > <pause milliseconds="10000"/> > <nop> > <action> > <exec play_pcap_audio="pcap/mod_01012013_240to21.pcap"/> > </action> > </nop> > > <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: field0 <sip:field0@[local_ip]:[local_port]>;tag=[call_number] > To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] > Call-ID: [call_id] > CSeq: 2 BYE > Contact: sip:field0@[local_ip]:[local_port] > Max-Forwards: 70 > Subject: Performance Test > Content-Length: 0 > ]]> > </send> > > <timewait milliseconds="1000"/> > <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/> > <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> > </scenario> > > Attached are Files. > > 1. Successful SIP Call PCAP sip_succ.pcap > 2. Failed SIPP call failed_sipp.pcap > 3. Scenario Used in SIPP crbt1.xml > > Best Regards, > Rajesh Bansal > > > On Mon, Jan 7, 2013 at 8:11 PM, Rob Day <r...@rkd.me.uk> wrote: >> >> On 7 January 2013 13:31, Rajesh Bansal <bansal.rajes...@gmail.com> wrote: >> > I need urgent help. I want to test ring back tone application in which >> > we >> > are playing media after 183(with SDP). But i am not able to make XML >> > (UAC) >> > scenario for the same. Kindly help me in this regard, >> >> Well - what problem are you having? What have you tried so far, what >> are you expecting it to do, and what is actually happening? >> >> (You may find that http://www.catb.org/esr/faqs/smart-questions.html >> helps you to make the best use of mailing lists such as these, >> particularly http://www.catb.org/esr/faqs/smart-questions.html#beprecise). >> >> >> Best, >> Rob > > ------------------------------------------------------------------------------ Master Visual Studio, SharePoint, SQL, ASP.NET, C# 2012, HTML5, CSS, MVC, Windows 8 Apps, JavaScript and much more. Keep your skills current with LearnDevNow - 3,200 step-by-step video tutorials by Microsoft MVPs and experts. SALE $99.99 this month only -- learn more at: http://p.sf.net/sfu/learnmore_122412 _______________________________________________ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
