Commits (6)
[MODULE_PARAMETERS]
# This section shall contain the values of all parameters that are defined in your TTCN-3 modules.
LibCommon_Time.PX_TAC := 30.0
LibCommon_Sync.PX_TSYNC_TIME_LIMIT := 30.0;
LibCommon_Sync.PX_TSHUT_DOWN_TIME_LIMIT := 30.0;
LibItsHttp_Pics.PICS_HEADER_HOST := "lis.gridgears.io" # Used for LIS
#LibItsHttp_Pics.PICS_HEADER_HOST := "ecrf.gridgears.io" # Used for ECRF
LibItsHttp_Pics.PICS_HEADER_CONTENT_TYPE := "application/held+xml;charset=utf-8"
#LibItsHttp_Pics.PICS_HEADER_CONTENT_TYPE := "application/lost+xml;charset=utf-8"
LibNg112_Pixits.PX_IMS_SUT_UE1_PUBLIC_USER := "alice-01"
LibNg112_Pixits.PX_IMS_SUT_UE1_HOME_DOMAIN := "plugtests.net"
LibNg112_Pixits.PX_IMS_TS_UE1_IPADDR := "conftest.plugtests.net" # Local UE address used for From header
LibNg112_Pixits.PX_IMS_SUT_UE1_BEARER_IPADDR := "conftest.plugtests.net" # Local UE address used for SDP connection attrubute
LibNg112_Pixits.PX_IMS_SUT_BCF_PORT := 5062
LibNg112_Pixits.PX_IMS_SUT_BCF_IPADDR := "127.0.0.1"
LibNg112_Pixits.PX_IMS_SUT_BCF_HOME_DOMAIN := "bcf22.sut.net"
AtsBCF_Pics.PICS_BCF_S_SIP_TCP1 := true # Set to true when using SIP over TCP ==> change SIP ports configuration
#LibNg112_Pixits.PX_IMS_TS_UE1_PORT := 5052 # SIP over TCP port
LibNg112_Pixits.PX_IMS_SUT_EMERGENCY_SERVICE := "sos"
LibNg112_Pixits.PX_IMS_SUT_EMERGENCY_HOME_DOMAIN := "psap1.frequentis.plugtests.net"
LibNg112_Pixits.PX_IMS_SUT_PCSCF1_IPADDR := "psap1.frequentis.plugtests.net"
LibNg112_Pixits.PX_IMS_SUT_UE_IPADDR := "psap1.frequentis.plugtests.net"
LibNg112_Pixits.PX_GEOLOCATION_ID := "//lis1.etsi.plugtests.net/locref?ref=660bc838-882e-4683-8241-af26bb5960d8"
AtsBCF_Pixits.PX_CALL_INFO_INCIDENT_ID := "<urn:emergency:uid:incidentid:56..3f:bcf.at>"
AtsBCF_Pixits.PX_CALL_INFO_CALL_ID := "<urn:emergency:uid:callid:a56e556d871:bcf.at>"
AtsBCF_Pixits.PX_CALL_INFO_SOURCE_ID := "<urn:emergency:uid:sourceid:a7231gc42:bcf.com>"
LibSip_PIXITS.PX_SIP_TRANSPORT := "UDP";
[LOGGING]
# In this section you can specify the name of the log file and the classes of events
# you want to log into the file or display on console (standard error).
LogFile := "../logs/AtsBCF/%e.%h-%r.%s"
FileMask := LOG_ALL | USER | DEBUG | MATCHING
ConsoleMask := LOG_ALL | USER | DEBUG | MATCHING
#FileMask := ERROR | WARNING | USER | MATCHING | EXECUTOR_RUNTIME | VERDICTOP
#ConsoleMask := ERROR | WARNING | USER | MATCHING | EXECUTOR_RUNTIME | VERDICTOP
LogSourceInfo := Stack
LogEntityName:= Yes
LogEventTypes:= Yes
#TimeStampFormat := DateTime
[TESTPORT_PARAMETERS]
# In this section you can specify parameters that are passed to Test Ports.
# SIP
# TCP
#Caller.SIPP.params := "SIP/TCP(debug=1,server=127.0.0.1,port=5060,use_ssl=0)"
#CallTalker.SIPP.params := "SIP/TCP(debug=1,server_mode=1,server=127.0.0.1,local_port=5080,use_ssl=0)"
# UDP
Caller.SIPP.params := "SIP/UDP(debug=1,dst_ip=127.0.0.1,dst_port=5060,src_port=5061,use_ssl=0)"
CallTalker.SIPP.params := "SIP/UDP(debug=1,server_mode=1,src_port=5080,reuse_incoming_source_adddress=1,use_ssl=0)"
[DEFINE]
# In this section you can create macro definitions,
# that can be used in other configuration file sections except [INCLUDE] and [ORDERED_INCLUDE].
[INCLUDE]
# To use configuration settings given in other configuration files,
# the configuration files just need to be listed in this section, with their full or relative pathnames.
[ORDERED_INCLUDE]
# To use configuration settings given in other configuration files,
# the configuration files just need to be listed in this section, with their full or relative pathnames.
[EXTERNAL_COMMANDS]
# This section can define external commands (shell scripts) to be executed by the ETS
# whenever a control part or test case is started or terminated.
#BeginTestCase := ""
#EndTestCase := ""
#BeginControlPart := ""
#EndControlPart := ""
[EXECUTE]
# In this section you can specify what parts of your test suite you want to execute.
AtsBCF_TestControl.control
# IUT forwards well-formated INVITE to PSAP
#AtsBCF_TestCases.TC_BCF_SIP_INVITE_BV_01
# IUT adds EES-source to the INVITE message
#AtsBCF_TestCases.TC_BCF_SIP_INVITE_BV_02
# IUT adds Incident Tracking ID within Call-Info header to the INVITE message.
#AtsBCF_TestCases.TC_BCF_SIP_INVITE_BV_03
# IUT adds Call ID within Call-Info header to the INVITE message.
#AtsBCF_TestCases.TC_BCF_SIP_INVITE_BV_04
# IUT adds Source ID within Call-Info header to the INVITE message.
#AtsBCF_TestCases.TC_BCF_SIP_INVITE_BV_05
# IUT forwards well-formated MESSAGE to PSAP
#AtsBCF_TestCases.TC_BCF_SIP_MESSAGE_BV_01
# IUT adds EES-source to the MESSAGE message
#AtsBCF_TestCases.TC_BCF_SIP_MESSAGE_BV_02
# IUT adds Incident Tracking ID within Call-Info header to the MESSAGE message.
#AtsBCF_TestCases.TC_BCF_SIP_MESSAGE_BV_03
# IUT adds Call ID within Call-Info header to the MESSAGE message.
#AtsBCF_TestCases.TC_BCF_SIP_MESSAGE_BV_04
# IUT adds Source ID within Call-Info header to the MESSAGE message.
#AtsBCF_TestCases.TC_BCF_SIP_MESSAGE_BV_05
[GROUPS]
# In this section you can specify groups of hosts. These groups can be used inside the
# [COMPONENTS] section to restrict the creation of certain PTCs to a given set of hosts.
[COMPONENTS]
# This section consists of rules restricting the location of created PTCs.
[MAIN_CONTROLLER]
# The options herein control the behavior of MC.
KillTimer := 10.0
LocalAddress := 127.0.0.1
TCPPort := 12000
NumHCs := 1
echo "Simulate BCF side."
echo "CallerNetwork(TS Titan) <--> BCF(SUT SIPP) <--> PSAP(TS Titan)"
rm *.log
sipp -nr -t u1 -trace_msg -trace_err -m 2 -sf CallerNetwork.xml 127.0.0.1:5060
<?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" rrs="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>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<pause milliseconds="1"/>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<pause milliseconds="1"/>
<send retrans="1000">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv request="ACK"
optional="false"
rtd="true"
crlf="true">
</recv>
<pause milliseconds="4000"/>
<!-- 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">
<action>
<exec command="sipp -nr -t u1 -trace_msg -m 2 -sf Psap.xml 127.0.0.1:5080"></exec>
</action>
</pause>
<!-- pause milliseconds="4000"/-->
<recv request="BYE" timeout = "10000">
</recv>
<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>
<?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 UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<!-- <send retrans="1000"> -->
<send>
<![CDATA[
INVITE urn:service:sos SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: Alice <sip:alice-01@plugtests.net>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@plugtests.net>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:alice-01@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Plugtests Test
Geolocation: <cid:gE4dNNthX4QcNzCv@dec112.app>;inserted_by="sip:bob-04@plugtests.net"
Geolocation-Routing: no
Accept: application/pidf+xml
Content-Type: multipart/mixed; boundary=qfpbntOwkOXuJWki
Content-Length: [len]
--qfpbntOwkOXuJWki
Content-Type: application/sdp
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
--qfpbntOwkOXuJWki
Content-Type: application/pidf+xml
Content-ID: <gE4dNNthX4QcNzCv@dec112.app>
<?xml version="1.0" encoding="UTF-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:gp="urn:ietf:params:xml:ns:pidf:geopriv10" xmlns:gbp="urn:ietf:params:xml:ns:pidf:geopriv10:basicPolicy" xmlns:cl="urn:ietf:params:xml:ns:pidf:geopriv10:civicAddr" xmlns:gml="http://www.opengis.net/gml" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" entity="pres:alice@atlanta.example.com"><dm:device id="01003118-0200-d09c-a762-00046e180003"><gp:geopriv><gp:location-info><gml:location><gml:Point srsName="urn:ogc:def:crs:EPSG::4326"><gml:pos>43.62303240 7.04618454</gml:pos></gml:Point></gml:location></gp:location-info><gp:usage-rules><gbp:retransmission-allowed>false</gbp:retransmission-allowed><gbp:retention-expiry>2018-04-16T08:23:31.036Z</gbp:retention-expiry></gp:usage-rules><gp:method>gps</gp:method></gp:geopriv><dm:deviceID>01003118-0200-d09c-a762-00046e180003</dm:deviceID><dm:timestamp>2018-04-15T08:23:31.036Z</dm:timestamp></dm:device></presence>
--qfpbntOwkOXuJWki--
]]>
</send>
<recv response="100"
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]@[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: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[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 milliseconds="15000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="1000">
<![CDATA[
BYE 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: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[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>
echo "Simulate BCF side."
echo "CallerNetwork(TS Titan) <--> BCF(SUT SIPP) <--> PSAP(TS Titan)"
rm *.log
sipp -nr -t u1 -trace_msg -trace_err -m 2 -sf CallerNetwork.xml 127.0.0.1:5060
<?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" rrs="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>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<pause milliseconds="1"/>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<pause milliseconds="1"/>
<send retrans="1000">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv request="ACK"
optional="false"
rtd="true"
crlf="true">
</recv>
<pause milliseconds="4000"/>
<!-- 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">
<action>
<exec command="sipp -nr -t u1 -trace_msg -m 2 -sf Psap.xml 127.0.0.1:5080"></exec>
</action>
</pause>
<pause milliseconds="500"/>
<recv request="BYE" timeout = "10000">
</recv>
<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>
<?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 UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<!-- <send retrans="1000"> -->
<send>
<![CDATA[
INVITE urn:service:sos SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: Alice <sip:alice-01@plugtests.net>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@plugtests.net>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:alice-01@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Plugtests Test
Geolocation: <cid:gE4dNNthX4QcNzCv@dec112.app>;inserted_by="sip:bob-04@plugtests.net"
Geolocation-Routing: no
Accept: application/pidf+xml
Content-Type: multipart/mixed; boundary=qfpbntOwkOXuJWki
Content-Length: [len]
--qfpbntOwkOXuJWki
Content-Type: application/sdp
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
--qfpbntOwkOXuJWki
Content-Type: application/pidf+xml
Content-ID: <gE4dNNthX4QcNzCv@dec112.app>
<?xml version="1.0" encoding="UTF-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:gp="urn:ietf:params:xml:ns:pidf:geopriv10" xmlns:gbp="urn:ietf:params:xml:ns:pidf:geopriv10:basicPolicy" xmlns:cl="urn:ietf:params:xml:ns:pidf:geopriv10:civicAddr" xmlns:gml="http://www.opengis.net/gml" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" entity="pres:alice@atlanta.example.com"><dm:device id="01003118-0200-d09c-a762-00046e180003"><gp:geopriv><gp:location-info><gml:location><gml:Point srsName="urn:ogc:def:crs:EPSG::4326"><gml:pos>43.62303240 7.04618454</gml:pos></gml:Point></gml:location></gp:location-info><gp:usage-rules><gbp:retransmission-allowed>false</gbp:retransmission-allowed><gbp:retention-expiry>2018-04-16T08:23:31.036Z</gbp:retention-expiry></gp:usage-rules><gp:method>gps</gp:method></gp:geopriv><dm:deviceID>01003118-0200-d09c-a762-00046e180003</dm:deviceID><dm:timestamp>2018-04-15T08:23:31.036Z</dm:timestamp></dm:device></presence>
--qfpbntOwkOXuJWki--
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" 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]@[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: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[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 milliseconds="15000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="1000">
<![CDATA[
BYE 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: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[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>
echo "Simulate BCF side."
echo "CallerNetwork(TS Titan) <--> BCF(SUT SIPP) <--> PSAP(TS Titan)"
rm *.log
sipp -nr -t u1 -trace_msg -trace_err -m 2 -sf CallerNetwork.xml 127.0.0.1:5060
<?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" rrs="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>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<pause milliseconds="1"/>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<pause milliseconds="1"/>
<send retrans="1000">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv request="ACK"
optional="false"
rtd="true"
crlf="true">
</recv>
<pause milliseconds="4000"/>
<!-- 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">
<action>
<exec command="sipp -nr -t u1 -trace_msg -m 2 -sf Psap.xml 127.0.0.1:5080"></exec>
</action>
</pause>
<!-- pause milliseconds="4000"/-->
<recv request="BYE" timeout = "10000">
</recv>
<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>
<?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 UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<!-- <send retrans="1000"> -->
<send>
<![CDATA[
INVITE urn:service:sos SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: Alice <sip:alice-01@plugtests.net>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@plugtests.net>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:alice-01@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Plugtests Test
Call-Info: <urn:emergency:uid:incidentid:56..3f:bcf.at>;purpose=EmergencyCallData.IncidentId
Geolocation: <cid:gE4dNNthX4QcNzCv@dec112.app>;inserted_by="sip:bob-04@plugtests.net"
Geolocation-Routing: no
Accept: application/pidf+xml
Content-Type: multipart/mixed; boundary=qfpbntOwkOXuJWki
Content-Length: [len]
--qfpbntOwkOXuJWki
Content-Type: application/sdp
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
--qfpbntOwkOXuJWki
Content-Type: application/pidf+xml
Content-ID: <gE4dNNthX4QcNzCv@dec112.app>
<?xml version="1.0" encoding="UTF-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:gp="urn:ietf:params:xml:ns:pidf:geopriv10" xmlns:gbp="urn:ietf:params:xml:ns:pidf:geopriv10:basicPolicy" xmlns:cl="urn:ietf:params:xml:ns:pidf:geopriv10:civicAddr" xmlns:gml="http://www.opengis.net/gml" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" entity="pres:alice@atlanta.example.com"><dm:device id="01003118-0200-d09c-a762-00046e180003"><gp:geopriv><gp:location-info><gml:location><gml:Point srsName="urn:ogc:def:crs:EPSG::4326"><gml:pos>43.62303240 7.04618454</gml:pos></gml:Point></gml:location></gp:location-info><gp:usage-rules><gbp:retransmission-allowed>false</gbp:retransmission-allowed><gbp:retention-expiry>2018-04-16T08:23:31.036Z</gbp:retention-expiry></gp:usage-rules><gp:method>gps</gp:method></gp:geopriv><dm:deviceID>01003118-0200-d09c-a762-00046e180003</dm:deviceID><dm:timestamp>2018-04-15T08:23:31.036Z</dm:timestamp></dm:device></presence>
--qfpbntOwkOXuJWki--
]]>
</send>
<recv response="100"
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]@[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: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[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 milliseconds="15000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="1000">
<![CDATA[
BYE 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: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[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>
echo "Simulate BCF side."
echo "CallerNetwork(TS Titan) <--> BCF(SUT SIPP) <--> PSAP(TS Titan)"
rm *.log
sipp -nr -t u1 -trace_msg -trace_err -m 2 -sf CallerNetwork.xml 127.0.0.1:5060
<?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" rrs="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>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<pause milliseconds="1"/>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<pause milliseconds="1"/>
<send retrans="1000">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv request="ACK"
optional="false"
rtd="true"
crlf="true">
</recv>
<pause milliseconds="4000"/>
<!-- 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">
<action>
<exec command="sipp -nr -t u1 -trace_msg -m 2 -sf Psap.xml 127.0.0.1:5080"></exec>
</action>
</pause>
<!-- pause milliseconds="4000"/-->
<recv request="BYE" timeout = "10000">
</recv>
<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>
<?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 UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<!-- <send retrans="1000"> -->
<send>
<![CDATA[
INVITE urn:service:sos SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: Alice <sip:alice-01@plugtests.net>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@plugtests.net>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:alice-01@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Plugtests Test
Call-Info: <urn:emergency:uid:callid:a56e556d871:bcf.at>;purpose=EmergencyCallData.CallId
Geolocation: <cid:gE4dNNthX4QcNzCv@dec112.app>;inserted_by="sip:bob-04@plugtests.net"
Geolocation-Routing: no
Accept: application/pidf+xml
Content-Type: multipart/mixed; boundary=qfpbntOwkOXuJWki
Content-Length: [len]
--qfpbntOwkOXuJWki
Content-Type: application/sdp
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
--qfpbntOwkOXuJWki
Content-Type: application/pidf+xml
Content-ID: <gE4dNNthX4QcNzCv@dec112.app>
<?xml version="1.0" encoding="UTF-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:gp="urn:ietf:params:xml:ns:pidf:geopriv10" xmlns:gbp="urn:ietf:params:xml:ns:pidf:geopriv10:basicPolicy" xmlns:cl="urn:ietf:params:xml:ns:pidf:geopriv10:civicAddr" xmlns:gml="http://www.opengis.net/gml" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" entity="pres:alice@atlanta.example.com"><dm:device id="01003118-0200-d09c-a762-00046e180003"><gp:geopriv><gp:location-info><gml:location><gml:Point srsName="urn:ogc:def:crs:EPSG::4326"><gml:pos>43.62303240 7.04618454</gml:pos></gml:Point></gml:location></gp:location-info><gp:usage-rules><gbp:retransmission-allowed>false</gbp:retransmission-allowed><gbp:retention-expiry>2018-04-16T08:23:31.036Z</gbp:retention-expiry></gp:usage-rules><gp:method>gps</gp:method></gp:geopriv><dm:deviceID>01003118-0200-d09c-a762-00046e180003</dm:deviceID><dm:timestamp>2018-04-15T08:23:31.036Z</dm:timestamp></dm:device></presence>
--qfpbntOwkOXuJWki--
]]>
</send>
<recv response="100"
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]@[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: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[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 milliseconds="15000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="1000">
<![CDATA[
BYE 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: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[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>
echo "Simulate BCF side."
echo "CallerNetwork(TS Titan) <--> BCF(SUT SIPP) <--> PSAP(TS Titan)"
rm *.log
sipp -nr -t u1 -trace_msg -trace_err -m 2 -sf CallerNetwork.xml 127.0.0.1:5060
<?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" rrs="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>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<pause milliseconds="1"/>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<pause milliseconds="1"/>
<send retrans="1000">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv request="ACK"
optional="false"
rtd="true"
crlf="true">
</recv>
<pause milliseconds="4000"/>
<!-- 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">
<action>
<exec command="sipp -nr -t u1 -trace_msg -m 2 -sf Psap.xml 127.0.0.1:5080"></exec>
</action>
</pause>
<!-- pause milliseconds="4000"/-->
<recv request="BYE" timeout = "10000">
</recv>
<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>
<?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 UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<!-- <send retrans="1000"> -->
<send>
<![CDATA[
INVITE urn:service:sos SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: Alice <sip:alice-01@plugtests.net>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@plugtests.net>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:alice-01@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Plugtests Test
Call-Info: <urn:emergency:uid:sourceid:a7231gc42:bcf.com>;purpose=EmergencyCallData.SourceId
Geolocation: <cid:gE4dNNthX4QcNzCv@dec112.app>;inserted_by="sip:bob-04@plugtests.net"
Geolocation-Routing: no
Accept: application/pidf+xml
Content-Type: multipart/mixed; boundary=qfpbntOwkOXuJWki
Content-Length: [len]
--qfpbntOwkOXuJWki
Content-Type: application/sdp
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
--qfpbntOwkOXuJWki
Content-Type: application/pidf+xml
Content-ID: <gE4dNNthX4QcNzCv@dec112.app>
<?xml version="1.0" encoding="UTF-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:gp="urn:ietf:params:xml:ns:pidf:geopriv10" xmlns:gbp="urn:ietf:params:xml:ns:pidf:geopriv10:basicPolicy" xmlns:cl="urn:ietf:params:xml:ns:pidf:geopriv10:civicAddr" xmlns:gml="http://www.opengis.net/gml" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" entity="pres:alice@atlanta.example.com"><dm:device id="01003118-0200-d09c-a762-00046e180003"><gp:geopriv><gp:location-info><gml:location><gml:Point srsName="urn:ogc:def:crs:EPSG::4326"><gml:pos>43.62303240 7.04618454</gml:pos></gml:Point></gml:location></gp:location-info><gp:usage-rules><gbp:retransmission-allowed>false</gbp:retransmission-allowed><gbp:retention-expiry>2018-04-16T08:23:31.036Z</gbp:retention-expiry></gp:usage-rules><gp:method>gps</gp:method></gp:geopriv><dm:deviceID>01003118-0200-d09c-a762-00046e180003</dm:deviceID><dm:timestamp>2018-04-15T08:23:31.036Z</dm:timestamp></dm:device></presence>
--qfpbntOwkOXuJWki--
]]>
</send>
<recv response="100"
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]@[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: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[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 milliseconds="15000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="1000">
<![CDATA[
BYE 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: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[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>
echo "Simulate BCF side."
echo "CallerNetwork(TS Titan) <--> BCF(SUT SIPP) <--> PSAP(TS Titan)"
rm *.log
sipp -nr -t u1 -trace_msg -trace_err -m 2 -sf CallerNetwork.xml 127.0.0.1:5060
<?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="MESSAGE" crlf="true" rrs="true"> -->
<recv request="MESSAGE">
</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. -->
<pause milliseconds="1"/>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
]]>
</send>
<pause milliseconds="4000"/>
<!-- 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">
<action>
<exec command="sipp -nr -t u1 -trace_msg -m 2 -sf Psap.xml 127.0.0.1:5080"></exec>
</action>
</pause>
<pause milliseconds="8000"/>
<recv response="200"></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>
<?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 UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="1000">
<![CDATA[
MESSAGE urn:service:sos SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: Alice <sip:alice-01@plugtests.net>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@plugtests.net>
Call-ID: [call_id]
CSeq: 1 MESSAGE
Contact: sip:alice-01@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Plugtests Test
Geolocation: <cid:gE4dNNthX4QcNzCv@dec112.app>;inserted_by="sip:bob-04@plugtests.net"
Geolocation-Routing: no
]]>
</send>
<!-- 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>
<!-- 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>
echo "Simulate BCF side."
echo "CallerNetwork(TS Titan) <--> BCF(SUT SIPP) <--> PSAP(TS Titan)"
rm *.log
sipp -nr -t u1 -trace_msg -trace_err -m 2 -sf CallerNetwork.xml 127.0.0.1:5060