Validate TC_PSAP_SIP_INVITE_BV_03 with SIPp (e3ae4571) · Commits · EMTEL - Emergency Communications / NG112 Conformance Test Suites

etc/AtsPSAP/AtsNg112_Frequentis_psap.cfg_

+3 −0

Original line number	Diff line number	Diff line
		@@ -28,6 +28,9 @@ AtsPSAP_Pics.PICS_PSAP_S_SIP_TCP1 := false # Set to true when usin
		AtsPSAP_Pics.PICS_PSAP_S_SIP_UDP1 := true
		AtsPSAP_Pics.PICS_PSAP_USE_TLS := true

		AtsPSAP_Pics.PICS_PSAP_B_SDP_ULA1 := false
		AtsPSAP_Pics.PICS_PSAP_B_SDP_ALA1 := true

		LibSip_PIXITS.PX_SIP_TRANSPORT := "UDP" # Transport is TCP or TCP

		[LOGGING]

mocks/AtsPSAP/TC_PSAP_DEQUEUE_BV_01/TC_PSAP_DEQUEUE_BV_01.json

0 → 100644

+72 −0

Original line number	Diff line number	Diff line
		{
		"uuid": "dafbe210-ba60-11ec-b294-23089ec390f3",
		"lastMigration": 19,
		"name": "PSAP",
		"endpointPrefix": "",
		"latency": 0,
		"port": 30044,
		"routes": [
		{
		"uuid": "dafbe211-ba60-11ec-b294-23089ec390f3",
		"documentation": "",
		"method": "get",
		"endpoint": "",
		"responses": [
		{
		"uuid": "dafbe212-ba60-11ec-b294-23089ec390f3",
		"body": "{\n \"DequeueRegistrationExpirationTime\": 10,\n \"DequeueRegistrationStatusCode\": 0\n}\n",
		"latency": 0,
		"statusCode": 200,
		"label": "",
		"headers": [
		{
		"key": "Content-Type",
		"value": "application/json;charset=utf-8"
		}
		],
		"filePath": "",
		"sendFileAsBody": false,
		"rules": [],
		"rulesOperator": "OR",
		"disableTemplating": false,
		"fallbackTo404": false
		}
		],
		"enabled": true,
		"randomResponse": false,
		"sequentialResponse": false
		}
		],
		"proxyMode": false,
		"proxyHost": "",
		"proxyRemovePrefix": false,
		"cors": true,
		"headers": [
		{
		"key": "Content-Type",
		"value": "application/json"
		}
		],
		"proxyReqHeaders": [
		{
		"key": "",
		"value": ""
		}
		],
		"proxyResHeaders": [
		{
		"key": "",
		"value": ""
		}
		],
		"hostname": "0.0.0.0",
		"tlsOptions": {
		"enabled": false,
		"type": "CERT",
		"pfxPath": "",
		"certPath": "",
		"keyPath": "",
		"caPath": "",
		"passphrase": ""
		}
		}
		No newline at end of file

mocks/AtsPSAP/TC_PSAP_SIP_INVITE_BV_02/uac_urn.xml→mocks/AtsPSAP/TC_PSAP_SIP_INVITE_BV_03/PsapCallTalker.xml

+17 −4

Original line number	Diff line number	Diff line
		@@ -27,23 +27,36 @@

		INVITE urn:service:sos SIP/2.0
		Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
		From: Alice <sip:alice-05@plugtests.net>;tag=[pid]SIPpTag00[call_number]
		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
		Content-Type: application/sdp
		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
		a=rtpmap:0 PCMA/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>
		@@ -85,7 +98,7 @@

		<!-- This delay can be customized by the -d command-line option -->
		<!-- or by adding a 'milliseconds = "value"' option here. -->
		<pause milliseconds="4000"/>
		<pause milliseconds="15000"/>

		<!-- The 'crlf' option inserts a blank line in the statistics report. -->
		<send retrans="500">

mocks/AtsPSAP/TC_PSAP_SIP_INVITE_BV_03/PsapCaller.sh

0 → 100755

+3 −0

Original line number	Diff line number	Diff line
		echo "Simulate PSAP Caller side (UAS)"
		rm *.log
		sipp -t u1 -trace_msg -trace_err -m 2 -sf PsapCaller.xml 127.0.0.1:5060

mocks/AtsPSAP/TC_PSAP_SIP_INVITE_BV_03/PsapCaller.xml

0 → 100644

+140 −0

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="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="500">
		<![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 PCMA/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 1 -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>

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