diff options
| author | Nanang Izzuddin <nanang@teluu.com> | 2009-11-05 13:33:18 +0000 |
|---|---|---|
| committer | Nanang Izzuddin <nanang@teluu.com> | 2009-11-05 13:33:18 +0000 |
| commit | 1f67de36d13cf9268d518b035bb3b9bd09e5b923 (patch) | |
| tree | 3d51ec16548de5b68c66a5d4b8a57fa0338f4df4 | |
| parent | 57225e0388a5368db07f53a254216ba26ade8ec6 (diff) | |
Ticket #954: Added sipp test scenario for issue 1 (bad SE in 200 response, less than min-SE in request).
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2987 74dad513-b988-da41-8d7b-12977e46ad98
| -rw-r--r-- | tests/pjsua/scripts-sipp/uas-422-then-200-bad-se.xml | 119 |
1 files changed, 119 insertions, 0 deletions
diff --git a/tests/pjsua/scripts-sipp/uas-422-then-200-bad-se.xml b/tests/pjsua/scripts-sipp/uas-422-then-200-bad-se.xml new file mode 100644 index 00000000..537f242b --- /dev/null +++ b/tests/pjsua/scripts-sipp/uas-422-then-200-bad-se.xml @@ -0,0 +1,119 @@ +<?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 'uas' scenario. -->
+<!-- -->
+
+<scenario name="Basic UAS responder">
+ <!-- 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 request="INVITE" crlf="true">
+ </recv>
+
+ <!-- The '[last_*]' keyword is replaced automatically by the -->
+ <!-- specified header if it was present in the last message received -->
+ <!-- (except if it was a retransmission). If the header was not -->
+ <!-- present or if no message has been received, the '[last_*]' -->
+ <!-- keyword is discarded, and all bytes until the end of the line -->
+ <!-- are also discarded. -->
+ <!-- -->
+ <!-- If the specified header was present several times in the -->
+ <!-- message, all occurences are concatenated (CRLF seperated) -->
+ <!-- to be used in place of the '[last_*]' keyword. -->
+
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 422 Session Timer too small
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Min-SE: 5400
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="ACK"
+ optional="true"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+
+ <recv request="INVITE" crlf="true">
+ </recv>
+
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
+ Allow-Events: telephone-event
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Supported: replaces
+ Session-Expires: 3600;refresher=uas
+ Require: timer
+ Content-Type: application/sdp
+ Content-Disposition: session;handling=required
+ Content-Length: [len]
+
+ v=0
+ o=Some-UserAgent 68 210 IN IP4 [local_ip]
+ s=SIP Call
+ c=IN IP4 [local_ip]
+ t=0 0
+ m=audio 17294 RTP/AVP 18 101
+ c=IN IP4 [local_ip]
+ a=rtpmap:18 G729/8000
+ a=fmtp:18 annexb=no
+ a=rtpmap:101 telephone-event/8000
+ a=fmtp:101 0-16
+ a=ptime:20
+
+ ]]>
+ </send>
+
+ <recv request="ACK"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+
+ <!-- Keep the call open for a while in case the 200 is lost to be -->
+ <!-- able to retransmit it if we receive the BYE again. -->
+ <pause milliseconds="4000"/>
+
+
+ <!-- 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>
+
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