Validate TC_PSAP_SIP_INVITE_BV_06 with SIPp (7dda161b) · Commits · EMTEL - Emergency Communications / NG112 Conformance Test Suites

mocks/AtsPSAP/TC_PSAP_SIP_INVITE_BV_06/PsapCallTalker.xml

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		<?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

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		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

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		<?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">
		</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

+14 −4

Original line number	Diff line number	Diff line
		@@ -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);

ttcn/AtsPSAP/AtsPSAP_TestControl.ttcn

+4 −4

Original line number	Diff line number	Diff line
		@@ -25,14 +25,14 @@ module AtsPSAP_TestControl {
		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)); }
		}
		if (PICS_PSAP_S_SIP_UDP1 and PICS_PSAP_E_SIP_BSC1) {
		/*if (PICS_PSAP_S_SIP_UDP1 and PICS_PSAP_E_SIP_BSC1) {
		if (PICS_PSAP_B_SDP_ULA1) { execute(TC_PSAP_SIP_INVITE_BV_07(v_cSeq)); }
		}
		if (PICS_PSAP_E_SIP_URN1) {

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