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.jenkins.sh
View file @ dc3c9f7f
......@@ -22,6 +22,11 @@ fi
./run.sh
if [ "$1" == "--remove" ]
then
docker rmi --force etsiforge/ttf_t009_ng112:latest
fi
cd $run_dir
exit 0
etc/AtsPSAP/AtsNg112_Frequentis_psap.cfg_
View file @ dc3c9f7f
......@@ -23,9 +23,9 @@ LibNg112_Pixits.PX_IMS_SUT_PCSCF1_IPADDR := "psap1.frequentis.plugtests.
LibNg112_Pixits.PX_IMS_SUT_UE_IPADDR := "psap1.frequentis.plugtests.net"
AtsPSAP_Pics.PICS_PSAP_S_SIP_REGISTRATION := false
AtsPSAP_Pics.PICS_PSAP_S_SIP_TCP1 := true # Set to true when using SIP over TCP ==> change SIP ports configuration
AtsPSAP_Pics.PICS_PSAP_S_SIP_TCP1 := false # Set to true when using SIP over TCP ==> change SIP ports configuration
# Update LibSip_PIXITS.PX_SIP_TRANSPORT according to true/false value
AtsPSAP_Pics.PICS_PSAP_S_SIP_UDP1 := false
AtsPSAP_Pics.PICS_PSAP_S_SIP_UDP1 := true
AtsPSAP_Pics.PICS_PSAP_USE_TLS := true
AtsPSAP_Pics.PICS_PSAP_B_SDP_ULA1 := true
......@@ -52,11 +52,11 @@ LogEventTypes:= Yes
# 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)"
#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)"
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)"
# ESRP
system.httpPort.params := "HTTP(codecs=json:json_codec)/TCP(debug=1,server=192.168.1.39,port=30044,use_ssl=0)"
......
mocks/AtsPSAP/TC_PSAP_SIP_INFO_BV_01/PsapCaller.xml
View file @ dc3c9f7f
......@@ -135,7 +135,7 @@
<pause milliseconds="4000"/>
<recv request="BYE">
<recv request="BYE" timeout="10000">
</recv>
<send>
......
mocks/AtsPSAP/TC_PSAP_SIP_INVITE_BV_01/PsapCaller.xml
View file @ dc3c9f7f
......@@ -113,7 +113,7 @@
<pause milliseconds="4000"/>
<recv request="BYE">
<recv request="BYE" timeout="10000"/>
</recv>
<send>
......
mocks/AtsPSAP/TC_PSAP_SIP_INVITE_BV_02/PsapCaller_register.xml
View file @ dc3c9f7f
......@@ -67,7 +67,7 @@
<pause milliseconds="12000"/>
<recv request="BYE">
<recv request="BYE" timeout="10000">
</recv>
<send>
......
mocks/AtsPSAP/TC_PSAP_SIP_INVITE_BV_03/PsapCaller.xml
View file @ dc3c9f7f
......@@ -113,7 +113,7 @@
<pause milliseconds="4000"/>
<recv request="BYE">
<recv request="BYE" timeout="10000">
</recv>
<send>
......
mocks/AtsPSAP/TC_PSAP_SIP_INVITE_BV_04/PsapCaller_register.xml
View file @ dc3c9f7f
......@@ -67,7 +67,7 @@
<pause milliseconds="12000"/>
<recv request="BYE">
<recv request="BYE" timeout="10000">
</recv>
<send>
......
mocks/AtsPSAP/TC_PSAP_SIP_INVITE_BV_05/PsapCaller.xml
View file @ dc3c9f7f
......@@ -113,7 +113,7 @@
<pause milliseconds="4000"/>
<recv request="BYE">
<recv request="BYE" timeout="10000">
</recv>
<send>
......
mocks/AtsPSAP/TC_PSAP_SIP_INVITE_BV_06/PsapCallTalker.xml 0 → 100644
View file @ dc3c9f7f
<?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[
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>
mocks/AtsPSAP/TC_PSAP_SIP_INVITE_BV_06/PsapCaller.sh 0 → 100755
View file @ dc3c9f7f
echo "Simulate PSAP Caller side (UAS)"
rm *.log
sipp -t t1 -trace_msg -trace_calldebug -trace_err -m 2 -sf PsapCaller.xml 127.0.0.1:5060
mocks/AtsPSAP/TC_PSAP_SIP_INVITE_BV_06/PsapCaller.xml 0 → 100644
View file @ dc3c9f7f
<?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 -t t1 -trace_msg -m 2 -sf PsapCallTalker.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>
mocks/AtsPSAP/TC_PSAP_SIP_INVITE_BV_08/PsapCallTalker.xml 0 → 100644
View file @ dc3c9f7f
<?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[
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>
mocks/AtsPSAP/TC_PSAP_SIP_INVITE_BV_08/PsapCaller.sh 0 → 100755
View file @ dc3c9f7f
echo "Simulate PSAP Caller side (UAS)"
rm *.log
sipp -t t1 -trace_msg -trace_calldebug -trace_err -m 2 -sf PsapCaller.xml 127.0.0.1:5060
mocks/AtsPSAP/TC_PSAP_SIP_INVITE_BV_08/PsapCaller.xml 0 → 100644
View file @ dc3c9f7f
<?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 -t t1 -trace_msg -m 2 -sf PsapCallTalker.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>
ttcn/AtsPSAP/AtsPSAP_TestCases.ttcn
View file @ dc3c9f7f
......@@ -950,12 +950,17 @@ module AtsPSAP_TestCases {
);
v_invite.msgHeader.contentType := m_contentType(m_mimeMultipart);
f_SendINVITE(v_invite);
f_awaitingResponseIgnore100Trying(mw_Response_Base(c_statusLine200, vc_callId, vc_cSeq));
// awaiting 180 RINGING
f_awaitingResponseIgnore100Trying(mw_Response_Base(c_statusLine180, vc_callId, vc_cSeq));
// awaiting 200 OK INVITE
f_awaitingResponse(mw_Response_Base(c_statusLine200, vc_callId, vc_cSeq));
// Send INVITE ACK
LibSip_Steps.f_setHeadersACK();
f_SendACK(m_ACK_Request_route(vc_requestUri, vc_callId, vc_response.msgHeader.cSeq, vc_from, vc_to, vc_via, vc_route));
f_selfOrClientSyncAndVerdictPreamble(c_prDone, e_success);
// Test Body
f_sleep(10.0);
f_check_Conversation(PX_CHECK_CONVERSATION);
f_selfOrClientSyncAndVerdictTestBody(c_tbDone, e_success);
......@@ -979,15 +984,20 @@ module AtsPSAP_TestCases {
// Test adapter configuration
// Preamble
f_IMS_preamble_woRegistration(p_cSeq_s)
LibIms_Steps.f_setHeadersINVITE(p_cSeq_s, f_initSipUrl(c_serviceProfile_EMERGENCY), f_initSipUrl(c_userProfile_UE1atSUThome));
f_IMS_preamble_woRegistration(p_cSeq_s);
LibIms_Steps.f_setHeadersINVITE(p_cSeq_s, f_initUrnUrl("service", PX_IMS_SUT_EMERGENCY_SERVICE), f_initSipUrl(c_userProfile_UE1atSUThome));
f_selfOrClientSyncAndVerdictPreamble(c_prDone, e_success);
// Test Body
// Await INVITE
f_awaitingINVITE(mw_INVITE_Request_RequestURI(vc_requestUri));
// Send 180 RINGING
f_sendResponse(m_Response_18XonINVITE_UE(c_statusLine180, vc_callId, vc_cSeq, vc_caller_From, vc_caller_To, vc_via,
vc_contact));
// Send 200 OK INVITE
f_sendResponse(m_Response_2xxonINVITE_UE(c_statusLine200, vc_callId, vc_cSeq, vc_caller_From, vc_caller_To, vc_via,
vc_contact, f_recordroute(), valueof(m_MBody_SDP(vc_sdp_local))));
// Await ACK INVITE
f_awaitingACK(mw_ACK_Request_Base(vc_callId));
f_check_Conversation(PX_CHECK_CONVERSATION);
......@@ -1168,6 +1178,9 @@ module AtsPSAP_TestCases {
) runs on ImsComponent {
// Local variables
var template INVITE_Request v_invite;
var XSD.String v_point;
var XSD.String v_location_info;
var XSD.String v_status := "manual";
// Test control
......@@ -1179,14 +1192,11 @@ module AtsPSAP_TestCases {
// Test adapter configuration
// Preamble
//v_point := encvalue_unichar(valueof(m_point(PX_PRESENCE_USER_POS, PX_SRS_NAME)));
//v_location_info := encvalue_unichar(valueof(m_geopriv(m_location_info({ point := m_point(PX_PRESENCE_USER_POS, PX_SRS_NAME) }), m_LocPolicyType)));
v_location_info := encvalue_unichar(valueof(m_geopriv({ point := m_point(PX_PRESENCE_USER_POS, PX_SRS_NAME) }, m_LocPolicyType)));
f_IMS_preamble_woRegistration(p_cSeq_s);
//await REGISTRATION
f_awaitingREGISTER(mw_REGISTER_unauthorizedRequest_IMS);
// 200OK to complete the request
f_sendResponse(m_Response_2xxonREGISTER_IMS(c_statusLine200,vc_callId, vc_cSeq,vc_callee_From, vc_callee_To, vc_via,vc_contact,f_initSipUrl(c_interfaceProfile_IMS_SUT_PCSCF1),vc_caller_From.addressField.nameAddr.addrSpec));
LibIms_Steps.f_setHeadersINVITE(vc_cSeq, f_initSipUrl(c_serviceProfile_EMERGENCY), f_initSipUrl(c_userProfile_UE1atSUThome));
LibIms_Steps.f_setHeadersINVITE(p_cSeq_s, f_initSipUrl(c_serviceProfile_EMERGENCY), f_initSipUrl(c_userProfile_UE1atSUThome));
f_init_sdp_userprofile(vc_sdp_local, c_userProfile_UE1atSUThome, c_userProfile_Huawei);
v_invite := m_INVITE_Request_IMS_Ng112(vc_requestUri,
vc_callId,
......@@ -1199,28 +1209,35 @@ module AtsPSAP_TestCases {
-, -,
m_Supported_100rel,
-,
m_MBody_SDP(vc_sdp_local),
/*m_MBody_MIMESdpXml_Emergency(
m_MBody_MIMESdpXml_Emergency(
vc_sdp_local,
m_XmlBody_Presence(
m_presence(
PX_PRESENCE_USER,
{ m_tuple(
"fclnzj22sbtoxpj",
{ basic := omit, elem_list := { v_location_info } }
)
}
{ //tuple_list
m_tuple(
"0ehvxv4kxwrzzbp",
{
omit,
m_geopriv({ point := m_point(PX_PRESENCE_USER_POS, PX_SRS_NAME) }, m_LocPolicyType),
{ }
}
)
}
)
)
),*/
//TODO make geolocation as PIXIT and update MIME body and Content-Type header as multipart/mixed
m_geolocation({ scheme:= "https", components := {other := PX_GEOLOCATION_ID }, urlParameters := omit, headers := omit }, omit), // TODO Add a LIS component to do a request for location by reference
),
m_geolocation({ scheme:= "http", components := { other := PX_GEOLOCATION_ID }, urlParameters := omit, headers := omit }, omit),
m_geolocation_routing,
{ fieldName := USER_AGENT_E, userAgentBody := { "ETSI_Conf_testing" } }
);
//v_invite.msgHeader.contentType := m_contentType(m_mimeMultipart);
v_invite.msgHeader.contentType := m_contentType(m_mimeMultipart);
f_SendINVITE(v_invite);
f_awaitingResponseIgnore100Trying(mw_Response_Base(c_statusLine200, vc_callId, vc_cSeq));
// awaiting 180 RINGING
f_awaitingResponseIgnore100Trying(mw_Response_Base(c_statusLine180, vc_callId, vc_cSeq));
// awaiting 200 OK INVITE
f_awaitingResponse(mw_Response_Base(c_statusLine200, vc_callId, vc_cSeq));
// Send INVITE ACK
LibSip_Steps.f_setHeadersACK();
f_SendACK(m_ACK_Request_route(vc_requestUri, vc_callId, vc_response.msgHeader.cSeq, vc_from, vc_to, vc_via, vc_route));
f_selfOrClientSyncAndVerdictPreamble(c_prDone, e_success);
......@@ -1239,7 +1256,6 @@ module AtsPSAP_TestCases {
in CSeq p_cSeq_s
) runs on ImsComponent {
// Local variables
var template REGISTER_Request v_register;
// Test control
f_init_userprofile(c_userProfile_UE1atSUThome);
......@@ -1250,19 +1266,21 @@ module AtsPSAP_TestCases {
// Test adapter configuration
// Preamble with REGISTRATION
v_register := valueof(m_REGISTER_Request_Base(vc_requestUri, vc_callId, p_cSeq_s, vc_from, vc_to, vc_via_REG, vc_contact, vc_authorization));
f_IMS_preamble_withRegistration(p_cSeq_s, v_register);
//set Invite headers
LibIms_Steps.f_setHeadersINVITE(p_cSeq_s, f_initSipUrl(c_serviceProfile_EMERGENCY), f_initSipUrl(c_userProfile_UE1atSUThome));
// Preamble
f_IMS_preamble_woRegistration(p_cSeq_s);
LibIms_Steps.f_setHeadersINVITE(p_cSeq_s, f_initUrnUrl("service", PX_IMS_SUT_EMERGENCY_SERVICE), f_initSipUrl(c_userProfile_UE1atSUThome));
f_selfOrClientSyncAndVerdictPreamble(c_prDone, e_success);
// Test Body
// Await INVITE
f_awaitingINVITE(mw_INVITE_Request_RequestURI(vc_requestUri));
// Send 180 RINGING
f_sendResponse(m_Response_18XonINVITE_UE(c_statusLine180, vc_callId, vc_cSeq, vc_caller_From, vc_caller_To, vc_via,
vc_contact));
// Send 200 OK INVITE
f_sendResponse(m_Response_2xxonINVITE_UE(c_statusLine200, vc_callId, vc_cSeq, vc_caller_From, vc_caller_To, vc_via,
vc_contact, f_recordroute(), valueof(m_MBody_SDP(vc_sdp_local))));
// Await ACK INVITE
f_awaitingACK(mw_ACK_Request_Base(vc_callId));
f_check_Conversation(PX_CHECK_CONVERSATION);
......
ttcn/AtsPSAP/AtsPSAP_TestControl.ttcn
View file @ dc3c9f7f
......@@ -19,16 +19,16 @@ module AtsPSAP_TestControl {
}
//execute(TC_PSAP_SIP_INVITE_BV_02(v_cSeq));
/*if (PICS_PSAP_S_SIP_UDP1 and PICS_PSAP_E_SIP_URN3) {
if (PICS_PSAP_S_SIP_UDP1 and PICS_PSAP_E_SIP_URN3) {
if (PICS_PSAP_B_SDP_ULA1) { execute(TC_PSAP_SIP_INVITE_BV_01(v_cSeq)); }
if (PICS_PSAP_B_SDP_ALA1) { execute(TC_PSAP_SIP_INVITE_BV_03(v_cSeq)); }
if (PICS_PSAP_B_SDP_ALA1) { execute(TC_PSAP_SIP_INVITE_BV_04(v_cSeq)); }
}*/
//if (PICS_PSAP_B_SDP_ALA1) { execute(TC_PSAP_SIP_INVITE_BV_03(v_cSeq)); }
//if (PICS_PSAP_B_SDP_ALA1) { execute(TC_PSAP_SIP_INVITE_BV_04(v_cSeq)); }
}
//execute(TC_PSAP_SIP_INVITE_BV_04(v_cSeq));
if (PICS_PSAP_S_SIP_TCP1 and PICS_PSAP_E_SIP_URN3) {
/*if (PICS_PSAP_S_SIP_TCP1 and PICS_PSAP_E_SIP_URN3) {
if (PICS_PSAP_B_SDP_ULA1) { execute(TC_PSAP_SIP_INVITE_BV_05(v_cSeq)); }
}
/*if (PICS_PSAP_S_SIP_TCP1 and PICS_PSAP_E_SIP_URN1) {
if (PICS_PSAP_S_SIP_TCP1 and PICS_PSAP_E_SIP_URN1) {
if (PICS_PSAP_B_SDP_ULA1) { execute(TC_PSAP_SIP_INVITE_BV_06(v_cSeq)); }
//if (PICS_PSAP_B_SDP_ULA1) { execute(TC_PSAP_SIP_BYE_BV_01(v_cSeq)); }
}
......
virtualization/docker/Dockerfile
View file @ dc3c9f7f
FROM stfubuntu:18.04
MAINTAINER ETSI NG112
FROM stfubuntu:20.04
LABEL ETSI NG112
LABEL description="ETSI NG112 Docker Image"
ENV TERM=linux \
......
virtualization/docker/Dockerfile.stfubuntu
View file @ dc3c9f7f
FROM ubuntu:18.04
MAINTAINER ETSI STF
FROM ubuntu:20.04
LABEL ETSI STF
LABEL description="STF Docker Ubuntu image"
ENV TERM=linux \
......@@ -10,11 +9,8 @@ ENV TERM=linux \
RUN echo "docker-STF-dev" > /etc/hostname \
&& DEBIAN_FRONTEND=noninteractive apt-get update \
&& DEBIAN_FRONTEND=noninteractive apt-get dist-upgrade -y \
&& DEBIAN_FRONTEND=noninteractive apt-get install software-properties-common -y \
&& DEBIAN_FRONTEND=noninteractive add-apt-repository ppa:linuxuprising/java -y \
&& DEBIAN_FRONTEND=noninteractive apt-get update \
&& ( echo "oracle-java17-installer shared/accepted-oracle-license-v1-3 boolean true" | debconf-set-selections -v ) \
&& DEBIAN_FRONTEND=noninteractive apt-get --allow-unauthenticated install -y \
&& DEBIAN_FRONTEND=noninteractive apt-get install -y \
autoconf \
bison \
build-essential \
......@@ -25,10 +21,9 @@ RUN echo "docker-STF-dev" > /etc/hostname \
emacs \
expect \
flex \
g++-8 \
gcc-8 \
gdb \
git-core \
g++-9 \
gcc-9 \
git \
gnutls-bin \
graphviz \
iputils-ping \
......@@ -40,41 +35,27 @@ RUN echo "docker-STF-dev" > /etc/hostname \
libjsoncpp-dev \
libncurses5-dev \
libpcap-dev \
libqt5svg5-dev \
libssl-dev \
libtool-bin \
libtool \
libwireshark-dev \
libxml2-dev \
libxml2-utils \
libyaml-dev \
lsof \
net-tools \
ntp \
oracle-java17-installer \
oracle-java17-set-default \
openssh-server \
pkg-config \
python3-dev \
python3-pip \
python3-setuptools \
qt5-default \
qtmultimedia5-dev \
subversion \
sudo \
sshpass \
tcpdump \
texlive-font-utils \
texlive-latex-extra \
tshark \
tzdata \
valgrind \
vim \
vsftpd \
xutils-dev \
tree \
unzip \
wget \
xsltproc \
&& DEBIAN_FRONTEND=noninteractive apt-get autoremove --purge -y \
&& DEBIAN_FRONTEND=noninteractive apt-get autoclean \
......
virtualization/docker/build.sh
View file @ dc3c9f7f
......@@ -13,7 +13,7 @@ if [ "$1" == "--force-stfubuntu" ]; then
fi
# Check and build stfubuntu image
if [ -z `docker images -q stfubuntu` ]; then
docker build --no-cache --tag stfubuntu:18.04 -f Dockerfile.stfubuntu --force-rm . || exit 1
docker build --no-cache --tag stfubuntu:20.04 -f Dockerfile.stfubuntu --force-rm . || exit 1
fi
docker build --no-cache --tag ttf_t009_ng112 --force-rm . || ( echo "Docker build failed: $?"; exit 1 )
......
virtualization/docker/home/etc/init.d/20-eclipse.sh deleted 100755 → 0
View file @ dce25a4f
#!/bin/bash
#set -e
set -vx
echo -e "*****************************\n* Install eclipse\n*****************************\n"
ECLIPSE_URL='http://www.mirrorservice.org/sites/download.eclipse.org/eclipseMirror/technology/epp/downloads/release/2021-12/R/eclipse-cpp-2021-12-R-linux-gtk-x86_64.tar.gz'
TITAN_URL='https://www.eclipse.org/downloads/download.php?file=/titan/TITAN_Designer_and_Executor_plugin-8.1.0.zip'
mkdir -p ${HOME}/frameworks
cd ${HOME}/frameworks || exit 1
wget --progress=dot:mega $ECLIPSE_URL -Oeclipse-cpp-2020-06-R-linux-gtk-x86_64.tar.gz || exit 1
tar -zxvf ./eclipse-cpp-2020-06-R-linux-gtk-x86_64.tar.gz || exit 1
ln -sf ${HOME}/frameworks/eclipse/eclipse ${HOME}/bin/eclipse
rm -f ./eclipse-cpp-2020-06-R-linux-gtk-x86_64.tar.gz
if wget --progress=dot:mega $TITAN_URL -OTITAN_Designer_and_Executor_plugin.zip; then
eclipse/eclipse -noSplash -application org.eclipse.equinox.p2.director \
-destination /home/etsi/frameworks/eclipse \
-repository 'jar:file:///home/etsi/frameworks/TITAN_Designer_and_Executor_plugin.zip' \
-installIU TITAN_Designer.feature.group,TITAN_Executor.feature.group,TITAN_Log_Viewer.feature.group,Titan_external_dependencies.feature.group,Titanium.feature.group,Titanium_external_dependencies.feature.group
rm -f TITAN_Designer_and_Executor_plugin.zip
fi
cd ${HOME}