Hi all, I am testing a basic call scenario using the SIPP Script. I face an issue with this script. I see that my script is sending CANCEL message on receiving 180 Ringing.
On some analysis found that the other endpoint is sending 2 180 Ringing, hence included two <recv> tag for 180 ringing. Still CANCEL is getting sent, Can someone provide some hint on what could be the issue ? uac_sp_SIPP_list.xml --> SIPP Script Thanks Anisha
uac_sp_16200_errors.log
Description: uac_sp_16200_errors.log
uac_sp_10640_messages.log
Description: uac_sp_10640_messages.log
<?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 'uac' scenario with pcap (rtp) play -->
<!-- -->
<scenario name="UAC">
<!-- In client mod (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="0">
<![CDATA[
INVITE sip:ABCD@[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "Sipp Test"<sip:SippUser@[local_ip];user=phone>;tag=[call_number]
To: "VVX-600 Teleworker"<sip:ABCD@Y.Y.Y.Y>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: <sip:[local_ip]:[local_port];transport=udp>
Supported: 100rel
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, INFO, UPDATE, REFER, PRACK
Accept: application/media_control+xml, application/sdp, multipart/mixed
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: [len]
v=0
o=Broadworks 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
a=sendrecv
m=audio [auto_media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="180" optional="false">
</recv>
<recv response="200" rtd="true" crlf="true" rrs="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:ABCD@[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "Sipp Test"<sip:SippUser@[local_ip];user=phone>;tag=[call_number]
To: "VVX-600 Teleworker"<sip:ABCD@Y.Y.Y.Y:5060>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:SippUser@[local_ip]:[local_port]
Max-Forwards: 70
Content-Length: 0
]]>
</send>
<recv request="BYE"/>
<!-- 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>
<?xml version="1.0" encoding="ISO-8859-1" ?>
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