Commit 8a625752 authored by YannGarcia's avatar YannGarcia
Browse files

Update Makefile

parent 8cb55dee

Makefile

+4 −4
Original line number Diff line number Diff line
@@ -23,7 +23,7 @@ $$(foreach I, $$(includes), $$(eval all_includes += $(1)/$$(I))) 
$$(foreach M, $$(modules),  $$(eval $$(call IncludeModule, $$(if $$(filter /%, $$(M)), $$(M), $(1)/$$(M)))))

endef



all_includes := $(TTCN3_DIR)/include $(TTCN3_DIR)/src $(TOPDIR)/ccsrc/Framework/include

all_includes := $(TTCN3_DIR)/include $(TTCN3_DIR)/src

defines  += TITAN_RUNTIME_2 _NO_SOFTLINKS_ $(ATS) AS_USE_SSL

libs     += $(TTCN3_DIR)/lib/libttcn3-rt2-parallel.a -lstdc++fs
@@ -99,14 +99,14 @@ $(outdir) $(bindir): 
	mkdir -p $@



$(bindir)/$(ATS)$(EXE): $(gen_objects) $(cc_objects)

	g++ -g -O0 -o $@ $(LDFLAGS) $(gen_objects) $(cc_objects) $(libs)

	g++ -g -O0 -std=c++17 -o $@ $(LDFLAGS) $(gen_objects) $(cc_objects) $(libs)



$(gen_objects) :%.o :%.cc

	g++ -g -O0 -c $(addprefix -D, $(defines)) $(addprefix -I, $(includes)) -o $@ $<

	g++ -g -O0 -std=c++17 -c $(addprefix -D, $(defines)) $(addprefix -I, $(includes)) -o $@ $<



$(cc_objects) : $(outdir)/%.o : %.cc

	mkdir -p $(dir $@)

	g++ -g -O0 -c $(addprefix -D, $(defines)) $(addprefix -I, $(includes)) -o $@ $<

	g++ -g -O0 -std=c++17 -c $(addprefix -D, $(defines)) $(addprefix -I, $(includes)) -o $@ $<



$(gen_sources): $(outdir)/.generate

mocks/AtsPSAP/TC_PSAP_SIP_INVITE_BV_04/PsapCallTalker.xml~

deleted100644 → 0
+0 −131
Original line number Diff line number Diff line 
<?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="4000"/>



  <!-- 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_04/PsapCaller.xml~

deleted100644 → 0
+0 −122
Original line number Diff line number Diff line 
<?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="TC_PSAP_SIP_INVITE_BV_02_OUT_OF_CALL">

  <!-- 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 u1 -trace_msg -m 2 -sf PsapCallTalker.xml 127.0.0.1:5080"></exec>

    </action-->

  </pause>



  <pause milliseconds="4000"/>



  <!-- definition of the response time repartition table (unit is ms)   -->

  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>



  <!-- definition of the call length repartition table (unit is ms)     -->

  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>



</scenario>

scripts/devenv.bash.ubuntu

+5 −3
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@@ -59,11 +59,13 @@ then 
    export TTCN3_LICENSE_FILE=${TTCN3_DIR}/etc/license/license.dat

    export PATH_DEV_TTCN=${HOME}/dev/ttcn3

    # ITS support

    export PATH_DEV_ITS=${HOME}/dev/etsi_its

    export PATH_DEV_ITS=${HOME}/dev/TS.ITS

    # Emergency Communication support

    export PATH_DEV_EMCOM=${HOME}/dev/etsi_emcom

    export PATH_DEV_EMCOM=${HOME}/dev/NG112

    # Mobile-Edge Computing support

    export PATH_DEV_MEC=${HOME}/dev/etsi_mec

    export PATH_DEV_MEC=${HOME}/dev/gs032p3-ttcn-test-suite

    # CDM/CISE support

    export PATH_DEV_MEC=${HOME}/dev/cise

    # Validation folder

    export VALIDATION_DIR=${HOME}

fi

titan-test-system-framework @ a24f3748

Original line number Diff line number Diff line 
Subproject commit a49d8daee3e5df27198a1902a7c630a84351e427
 
Subproject commit a24f37489fd2fb50bfcd3a35b3e8e094a9c2bf6a