[asterisk-commits] mmichelson: branch mmichelson/phone-testsuite r3167 - /asterisk/team/mmichels...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Mar 29 15:56:54 CDT 2012
Author: mmichelson
Date: Thu Mar 29 15:56:50 2012
New Revision: 3167
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=3167
Log:
Add SIPp case that was for some reason not added previously.
Added:
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/non_digium_state_change/sipp/subscribe.xml (with props)
Added: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/non_digium_state_change/sipp/subscribe.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/non_digium_state_change/sipp/subscribe.xml?view=auto&rev=3167
==============================================================================
--- asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/non_digium_state_change/sipp/subscribe.xml (added)
+++ asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/non_digium_state_change/sipp/subscribe.xml Thu Mar 29 15:56:50 2012
@@ -1,0 +1,156 @@
+<?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[
+
+ SUBSCRIBE 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 SUBSCRIBE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Event: presence
+ Accept: application/pidf+xml
+ Expires: 3600
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ 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>
+
+ <!-- Initial NOTIFY upon subscribing -->
+ <recv request="NOTIFY" crlf="true" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- NOTIFY indicating ringing state -->
+ <recv request="NOTIFY" crlf="true" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- NOTIFY indicating answered state -->
+ <recv request="NOTIFY" crlf="true" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ SUBSCRIBE 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: [cseq] SUBSCRIBE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Event: presence
+ Accept: application/pidf+xml
+ Expires: 0
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true" />
+
+ <!-- NOTIFY terminating subscription -->
+ <recv request="NOTIFY" crlf="true" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- 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>
+
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/non_digium_state_change/sipp/subscribe.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/non_digium_state_change/sipp/subscribe.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/non_digium_state_change/sipp/subscribe.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
More information about the asterisk-commits
mailing list