[asterisk-commits] jrose: testsuite/asterisk/trunk r4801 - in /asterisk/trunk/tests/channels: SI...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Mar 7 09:58:14 CST 2014


Author: jrose
Date: Fri Mar  7 09:58:12 2014
New Revision: 4801

URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=4801
Log:
Testsuite: SIP hold tests with ICE for chan_sip and chan_pjsip

(closes issue ASTERISK-23213)
Reported by: Andrea Suisani
(closes issue ASTERISK-22911)
Reported by: Vytis Valentinavičius
Review: https://reviewboard.asterisk.org/r/3255/
Review: https://reviewboard.asterisk.org/r/3286/

Added:
    asterisk/trunk/tests/channels/SIP/sip_hold_ice/
    asterisk/trunk/tests/channels/SIP/sip_hold_ice/configs/
    asterisk/trunk/tests/channels/SIP/sip_hold_ice/configs/ast1/
    asterisk/trunk/tests/channels/SIP/sip_hold_ice/configs/ast1/extensions.conf   (with props)
    asterisk/trunk/tests/channels/SIP/sip_hold_ice/configs/ast1/rtp.conf   (with props)
    asterisk/trunk/tests/channels/SIP/sip_hold_ice/configs/ast1/sip.conf   (with props)
    asterisk/trunk/tests/channels/SIP/sip_hold_ice/run-test   (with props)
    asterisk/trunk/tests/channels/SIP/sip_hold_ice/sipp/
    asterisk/trunk/tests/channels/SIP/sip_hold_ice/sipp/inject_bridge.csv   (with props)
    asterisk/trunk/tests/channels/SIP/sip_hold_ice/sipp/phone_A.xml   (with props)
    asterisk/trunk/tests/channels/SIP/sip_hold_ice/sipp/phone_B.xml   (with props)
    asterisk/trunk/tests/channels/SIP/sip_hold_ice/test-config.yaml   (with props)
    asterisk/trunk/tests/channels/pjsip/hold_ice/
    asterisk/trunk/tests/channels/pjsip/hold_ice/configs/
    asterisk/trunk/tests/channels/pjsip/hold_ice/configs/ast1/
    asterisk/trunk/tests/channels/pjsip/hold_ice/configs/ast1/extensions.conf   (with props)
    asterisk/trunk/tests/channels/pjsip/hold_ice/configs/ast1/pjsip.conf   (with props)
    asterisk/trunk/tests/channels/pjsip/hold_ice/configs/ast1/rtp.conf   (with props)
    asterisk/trunk/tests/channels/pjsip/hold_ice/run-test   (with props)
    asterisk/trunk/tests/channels/pjsip/hold_ice/sipp/
    asterisk/trunk/tests/channels/pjsip/hold_ice/sipp/inject.csv   (with props)
    asterisk/trunk/tests/channels/pjsip/hold_ice/sipp/phone_A.xml   (with props)
    asterisk/trunk/tests/channels/pjsip/hold_ice/sipp/phone_B.xml   (with props)
    asterisk/trunk/tests/channels/pjsip/hold_ice/test-config.yaml   (with props)
Modified:
    asterisk/trunk/tests/channels/SIP/tests.yaml
    asterisk/trunk/tests/channels/pjsip/tests.yaml

Added: asterisk/trunk/tests/channels/SIP/sip_hold_ice/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold_ice/configs/ast1/extensions.conf?view=auto&rev=4801
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold_ice/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold_ice/configs/ast1/extensions.conf Fri Mar  7 09:58:12 2014
@@ -1,0 +1,7 @@
+[general]
+PHONE_TO_DIAL=SIP/phone_B
+
+[default]
+exten => basicdial,1,NoOp()
+	same => n,Dial(SIP/phone_B)
+	same => n,Hangup()

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Added: asterisk/trunk/tests/channels/SIP/sip_hold_ice/configs/ast1/rtp.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold_ice/configs/ast1/rtp.conf?view=auto&rev=4801
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold_ice/configs/ast1/rtp.conf (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold_ice/configs/ast1/rtp.conf Fri Mar  7 09:58:12 2014
@@ -1,0 +1,5 @@
+[general]
+rtpstart=10000
+rtpend=20000
+icesupport=yes
+stunaddr=stun.l.google.com:19302

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Added: asterisk/trunk/tests/channels/SIP/sip_hold_ice/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold_ice/configs/ast1/sip.conf?view=auto&rev=4801
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold_ice/configs/ast1/sip.conf (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold_ice/configs/ast1/sip.conf Fri Mar  7 09:58:12 2014
@@ -1,0 +1,29 @@
+[general]
+allowguest=no
+bindaddr=0.0.0.0
+sipdebug=yes
+directmedia=no
+
+[phone_A]
+type=friend
+context=default
+disallow=all
+allow=ulaw
+qualify=no
+insecure=invite
+host=127.0.0.2
+port=6080
+icesupport=yes
+avpf=yes
+
+[phone_B]
+type=friend
+context=default
+disallow=all
+allow=ulaw
+qualify=no
+insecure=invite
+host=127.0.0.3
+port=6081
+nat=no
+

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Added: asterisk/trunk/tests/channels/SIP/sip_hold_ice/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold_ice/run-test?view=auto&rev=4801
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold_ice/run-test (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold_ice/run-test Fri Mar  7 09:58:12 2014
@@ -1,0 +1,180 @@
+#!/usr/bin/env python
+"""
+Copyright (C) 2014, Digium, Inc.
+Jonathan Rose <jrose at digium.com>
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+"""
+
+import sys
+import logging
+
+sys.path.append("lib/python")
+
+from asterisk.test_case import TestCase
+from asterisk.sipp import SIPpScenario
+from twisted.internet import reactor
+from asterisk.version import AsteriskVersion
+
+LOGGER = logging.getLogger(__name__)
+INJECT_FILE_BRIDGE = "inject_bridge.csv"
+
+
+class SIPHold(TestCase):
+    """TestCase to execute and evaluate SIP hold/unhold with ICE scenario """
+
+    def __init__(self):
+        """ Constructor """
+        TestCase.__init__(self)
+        self.create_asterisk()
+
+        running_version = AsteriskVersion()
+
+        if (running_version < AsteriskVersion("12.0.0")):
+            #Pre-12
+            self.asterisk12Events = False
+        else:
+            self.asterisk12Events = True
+
+        self.sipp_phone_a_scenario = {'scenario': 'phone_A.xml',
+                                      '-i': '127.0.0.2',
+                                      '-p': ' 6080',
+                                      '-inf': INJECT_FILE_BRIDGE}
+        self.sipp_phone_b_scenario = {'scenario': 'phone_B.xml',
+                                      '-i': '127.0.0.3',
+                                      '-p': '6081',
+                                      '-inf': INJECT_FILE_BRIDGE}
+
+        self.moh_start_event = False
+        self.moh_stop_event = False
+        self.check_list_success_events = 0
+
+    def ami_connect(self, ami):
+        """ Reaction to new AMI connection
+
+        :param ami: AMI connection that was established
+        """
+        TestCase.ami_connect(self, ami)
+
+        ami.registerEvent('TestEvent', self.test_event_handler)
+
+        if self.asterisk12Events:
+            ami.registerEvent('MusicOnHoldStart', self.moh_start_event_handler)
+            ami.registerEvent('MusicOnHoldStop', self.moh_stop_event_handler)
+        else:
+            ami.registerEvent('MusicOnHold', self.moh_event_handler)
+
+        LOGGER.info("Starting SIP scenario")
+        self.execute_scenario()
+
+    def execute_scenario(self):
+        """Execute sipp scenarios and check results for a single test phase
+        """
+        sipp_a = SIPpScenario(self.test_name, self.sipp_phone_a_scenario)
+        sipp_b = SIPpScenario(self.test_name, self.sipp_phone_b_scenario)
+
+        # Start up the listener first - Phone A calls Phone B
+        sipp_b.run(self)
+        sipp_a.run(self)
+
+    def handle_checklist_create(self, ami, event):
+        """ Reacts to ICECHECKLISTCREATE user events
+
+        :param ami AMI connection the event was received from
+        :param event Event that was received
+        """
+        if event.get('result') == 'SUCCESS':
+            self.check_list_success_events += 1
+            LOGGER.debug("Received create check list success")
+            if self.check_list_success_events == 2 and \
+                    self.moh_stop_event:
+                self.evaluate_success()
+                self.stop_reactor()
+        else:
+            LOGGER.error("Failed to create check list - test failed")
+            self.set_passed(False)
+            self.stop_reactor()
+
+    def test_event_handler(self, ami, event):
+        """ Reacts to User Events and routes them based on their State Value
+
+        :param ami AMI connection the event was received from
+        :param event Event that was received
+        """
+        if event.get('state') == 'ICECHECKLISTCREATE':
+            self.handle_checklist_create(ami, event)
+
+    def moh_start_event_handler(self, ami, event):
+        """ Reacts to music on hold start events and tallies them.
+
+        :param ami: AMI connection the event was received from
+        :param event: Event that was received
+        """
+        LOGGER.debug("Received MOH start event")
+        self.moh_start_event = True
+
+    def moh_stop_event_handler(self, ami, event):
+        """ Reacts to music on hold stop events and tallies them.
+
+        :param ami: AMI connection the event was received from
+        :param event: Event that was received
+        """
+        LOGGER.debug("Received MOH stop event")
+        self.moh_stop_event = True
+        if self.check_list_success_events == 2:
+            self.evaluate_success()
+            self.stop_reactor()
+
+    def moh_event_handler(self, ami, event):
+        """ Reacts to MusicOnHold events (legacy events from Asterisk <= 11)
+
+        :param ami: AMI connection the event was received from
+        :param event: Event that was received
+        """
+        if event.get('state') == "Start":
+            LOGGER.debug("Received MOH start event")
+            self.moh_start_event = True
+        elif event.get('state') == "Stop":
+            LOGGER.debug("Received MOH stop event")
+            self.moh_stop_event = True
+            if self.check_list_success_events == 2:
+                self.evaluate_success()
+                self.stop_reactor()
+
+    def run(self):
+        """ Run the test and create an AMI connection """
+        TestCase.run(self)
+        self.create_ami_factory()
+
+    def evaluate_success(self):
+        """ Determine whether all of the conditions for passing the test
+            have been met and raise passed flag accordingly.
+        """
+        if not self.moh_start_event:
+            LOGGER.error("Failed to receive MOH start event")
+            self.set_passed(False)
+        if not self.moh_stop_event:
+            LOGGER.error("Failed to receive MOH stop event")
+            self.set_passed(False)
+        if not self.check_list_success_events == 2:
+            LOGGER.error("Expected 2 check list creation events. Got %d\n" %
+                         self.check_list_success_events)
+            self.set_passed(False)
+
+        self.set_passed(True)
+
+
+def main():
+    test = SIPHold()
+    reactor.run()
+    if test.passed:
+        return 0
+    return 1
+
+
+if __name__ == "__main__":
+    sys.exit(main())
+
+
+# vim:sw=4:ts=4:expandtab:textwidth=79

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Added: asterisk/trunk/tests/channels/SIP/sip_hold_ice/sipp/inject_bridge.csv
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold_ice/sipp/inject_bridge.csv?view=auto&rev=4801
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold_ice/sipp/inject_bridge.csv (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold_ice/sipp/inject_bridge.csv Fri Mar  7 09:58:12 2014
@@ -1,0 +1,2 @@
+SEQUENTIAL
+phone_A;phone_B;basicdial

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Added: asterisk/trunk/tests/channels/SIP/sip_hold_ice/sipp/phone_A.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold_ice/sipp/phone_A.xml?view=auto&rev=4801
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold_ice/sipp/phone_A.xml (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold_ice/sipp/phone_A.xml Fri Mar  7 09:58:12 2014
@@ -1,0 +1,222 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone A Hold with IP and Media Restrictions">
+
+	<send retrans="500">
+		<![CDATA[
+			INVITE sip:[field2]@[remote_ip]:[remote_port] SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
+			To: <sip:[field2]@[remote_ip]:[remote_port];user=phone>
+			CSeq: 1 INVITE
+			Call-ID: [call_id]
+			Contact: <sip:[field0]@[local_ip]:[local_port]>
+			Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Allow-Events: talk,hold,conference
+			Max-Forwards: 70
+			Content-Type: application/sdp
+			Content-Length: [len]
+
+			v=0
+			o=- 1324901698 1324901698 IN IP4 [local_ip]
+			s=Polycom IP Phone
+			c=IN IP4 [local_ip]
+			t=0 0
+			a=sendrecv
+			m=audio 2226 RTP/AVP 0 101
+			a=sendrecv
+			a=rtpmap:0 PCMU/8000
+			a=rtpmap:101 telephone-event/8000
+			a=candidate:474352566 1 udp 2113937151 10.24.16.82 58057 typ host generation 0
+			a=candidate:474352566 2 udp 2113937151 10.24.16.82 58057 typ host generation 0
+			a=candidate:3038348387 1 udp 1845501695 216.207.245.1 58057 typ srflx raddr 10.24.16.82 rport 58057 generation 0
+			a=candidate:3038348387 2 udp 1845501695 216.207.245.1 58057 typ srflx raddr 10.24.16.82 rport 58057 generation 0
+			a=candidate:1388705606 1 tcp 1509957375 10.24.16.82 0 typ host generation 0
+			a=candidate:1388705606 2 tcp 1509957375 10.24.16.82 0 typ host generation 0
+			a=ice-ufrag:/c7ZXBQs6LexKQPT
+			a=ice-pwd:pWzGkNwkFPRpcmVo7In+1Tnn
+			a=ice-options:google-ice
+		]]>
+	</send>
+
+	<recv response="100" optional="true" />
+
+	<recv response="180" optional="true" />
+
+	<recv response="183" optional="true" />
+
+	<recv response="200" />
+
+	<send>
+		<![CDATA[
+			ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: [field0] <sip:[field0]@[remote_ip]>;tag=[call_number]
+			To: <sip:[field1]@[remote_ip];user=phone>[peer_tag_param]
+			CSeq: 1 ACK
+			Call-ID: [call_id]
+			Contact: <sip:[field0]@[local_ip]:[local_port]>
+			Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Max-Forwards: 70
+			Content-Length: 0
+		]]>
+	</send>
+
+	<!-- Wait some period of time -->
+	<pause milliseconds="3000"/>
+
+	<!-- Modify RTP session to be send only -->
+	<send retrans="500">
+		<![CDATA[
+			INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
+			To: [field1] <sip:[field0]@[remote_ip]>
+			CSeq: [cseq] INVITE
+			Call-ID: [call_id]
+			Contact: <sip:[field0]@[local_ip]:[local_port]>
+			Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Supported: 100rel,replaces
+			Allow-Events: talk,hold,conference
+			Max-Forwards: 70
+			Content-Type: application/sdp
+			Content-Length: [len]
+
+			v=0
+			o=- 1325003603 1325003604 IN IP4 [local_ip]
+			s=Polycom IP Phone
+			c=IN IP4 [local_ip]
+			t=0 0
+			a=sendonly
+			m=audio 2226 RTP/AVP 0 101
+			a=sendonly
+			a=rtpmap:0 PCMU/8000
+			a=rtpmap:101 telephone-event/8000
+			a=candidate:474352566 1 udp 2113937151 10.24.16.82 58057 typ host generation 0
+			a=candidate:474352566 2 udp 2113937151 10.24.16.82 58057 typ host generation 0
+			a=candidate:3038348387 1 udp 1845501695 216.207.245.1 58057 typ srflx raddr 10.24.16.82 rport 58057 generation 0
+			a=candidate:3038348387 2 udp 1845501695 216.207.245.1 58057 typ srflx raddr 10.24.16.82 rport 58057 generation 0
+			a=candidate:1388705606 1 tcp 1509957375 10.24.16.82 0 typ host generation 0
+			a=candidate:1388705606 2 tcp 1509957375 10.24.16.82 0 typ host generation 0
+			a=ice-ufrag:/c7ZXBQs6LexKQPT
+			a=ice-pwd:pWzGkNwkFPRpcmVo7In+1Tnn
+			a=ice-options:google-ice
+		]]>
+	</send>
+
+	<recv response="100" optional="true" />
+
+	<recv response="200" />
+	<send>
+		<![CDATA[
+			ACK sip:[field2]@[remote_ip]:[remote_port] SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
+			To: <sip:[field2]@[remote_ip]>[peer_tag_param]
+			CSeq: [cseq] ACK
+			Call-ID: [call_id]
+			Contact: <sip:[field0]@[local_ip]:[local_port]>
+			Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Max-Forwards: 70
+			Content-Length: 0
+		]]>
+	</send>
+
+	<!-- Wait some period of time -->
+	<pause milliseconds="3000"/>
+
+	<!-- Unhold -->
+	<!-- Modify RTP session to be send/recv -->
+	<send retrans="500">
+		<![CDATA[
+			INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
+			To: [field1] <sip:[field0]@[remote_ip]>
+			CSeq: [cseq] INVITE
+			Call-ID: [call_id]
+			Contact: <sip:[field0]@[local_ip]:[local_port]>
+			Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Supported: 100rel,replaces
+			Allow-Events: talk,hold,conference
+			Max-Forwards: 70
+			Content-Type: application/sdp
+			Content-Length: [len]
+
+			v=0
+			o=- 1325003603 1325003605 IN IP4 [local_ip]
+			s=Polycom IP Phone
+			c=IN IP4 [local_ip]
+			t=0 0
+			a=sendrecv
+			m=audio 2226 RTP/AVP 0 101
+			a=sendrecv
+			a=rtpmap:0 PCMU/8000
+			a=rtpmap:101 telephone-event/8000
+			a=candidate:474352566 1 udp 2113937151 10.24.16.82 43995 typ host generation 0
+			a=candidate:474352566 2 udp 2113937151 10.24.16.82 43995 typ host generation 0
+			a=candidate:3038348387 1 udp 1845501695 216.207.245.1 43995 typ srflx raddr 10.24.16.82 rport 43995 generation 0
+			a=candidate:3038348387 2 udp 1845501695 216.207.245.1 43995 typ srflx raddr 10.24.16.82 rport 43995 generation 0
+			a=candidate:1388705606 1 tcp 1509957375 10.24.16.82 0 typ host generation 0
+			a=candidate:1388705606 2 tcp 1509957375 10.24.16.82 0 typ host generation 0
+			a=ice-ufrag:ZBrSbiG7KWV6Oxfs
+			a=ice-pwd:fPY7kNj+q4x2sn6zACkKURp+
+			a=ice-options:google-ice
+		]]>
+	</send>
+
+	<recv response="100" optional="true" />
+
+	<recv response="200" />
+
+	<send>
+		<![CDATA[
+			ACK sip:[field2]@[remote_ip]:[remote_port] SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
+			To: <sip:[field2]@[remote_ip]>[peer_tag_param]
+			CSeq: [cseq] ACK
+			Call-ID: [call_id]
+			Contact: <sip:[field0]@[local_ip]:[local_port]>
+			Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Max-Forwards: 70
+			Content-Length: 0
+		]]>
+	</send>
+
+	<!-- Wait some period of time -->
+	<pause milliseconds="1000"/>
+
+	<send>
+		<![CDATA[
+			BYE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
+			Via: SIP/2.0/UDP [local_ip]:[local_port];branch=[branch]
+			From: [field0] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+			To: [field1] <sip:[field1]@[remote_ip]>[peer_tag_param]
+			CSeq: [cseq] BYE
+			Call-ID: [call_id]
+			Contact: <sip:[field1]@[local_ip]:[local_port]>
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Max-Forwards: 70
+			Content-Length: 0
+		]]>
+	</send>
+
+	<recv response="200" />
+
+
+</scenario>

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Added: asterisk/trunk/tests/channels/SIP/sip_hold_ice/sipp/phone_B.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold_ice/sipp/phone_B.xml?view=auto&rev=4801
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold_ice/sipp/phone_B.xml (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold_ice/sipp/phone_B.xml Fri Mar  7 09:58:12 2014
@@ -1,0 +1,105 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone B Hold with Media Restrictions">
+
+	<recv request="INVITE" crlf="true" />
+
+	<send>
+		<![CDATA[
+			SIP/2.0 100 Trying
+			[last_Via:]
+			[last_From:]
+			[last_To:];tag=[call_number]
+			[last_Call-ID:]
+			[last_CSeq:]
+			Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Content-Length: 0
+		]]>
+	</send>
+
+	<send>
+		<![CDATA[
+			SIP/2.0 180 Ringing
+			[last_Via:]
+			[last_From:]
+			[last_To:];tag=[call_number]
+			[last_Call-ID:]
+			[last_CSeq:]
+			Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Allow-Events: talk,hold,conference
+			Accept-Language: en
+			Content-Length: 0
+		]]>
+	</send>
+
+	<pause milliseconds="200"/>
+
+	<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:[field1]@[local_ip]:[local_port];transport=[transport]>
+			Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+			Supported: 100rel,replaces
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Content-Type: application/sdp
+			Content-Length: [len]
+
+			v=0
+			o=- 1325003603 1325003604 IN IP4 [local_ip]
+			s=Polycom IP Phone
+			c=IN IP4 [local_ip]
+			t=0 0
+			a=sendrecv
+			m=audio 2226 RTP/AVP 0 101
+			a=sendrecv
+			a=rtpmap:0 PCMU/8000
+			a=rtpmap:101 telephone-event/8000
+		]]>
+	</send>
+
+	<!-- RECV ACK -->
+	<recv request="ACK"/>
+
+	<recv request="BYE"/>
+
+	<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:[field1]@[local_ip]:[local_port];transport=[transport]>
+			Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+			Supported: 100rel,replaces
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Content-Type: application/sdp
+			Content-Length: [len]
+
+			v=0
+			o=- 1324901698 1324901698 IN IP4 [local_ip]
+			s=Polycom IP Phone
+			c=IN IP4 [local_ip]
+			t=0 0
+			a=sendrecv
+			m=audio 2226 RTP/AVP 0 101
+			a=sendrecv
+			a=rtpmap:0 PCMU/8000
+			a=rtpmap:101 telephone-event/8000
+		]]>
+	</send>
+
+
+</scenario>

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Added: asterisk/trunk/tests/channels/SIP/sip_hold_ice/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold_ice/test-config.yaml?view=auto&rev=4801
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold_ice/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold_ice/test-config.yaml Fri Mar  7 09:58:12 2014
@@ -1,0 +1,22 @@
+testinfo:
+    summary: 'Test SIP Hold with ICE'
+    description: |
+        This tests a PJSIP assertion crash in 11.7 detailed somewhat in
+        https://issues.asterisk.org/jira/browse/ASTERISK-23213 - A call
+        is started with ICE enabled. A call is started from A to B with
+        some ICE candidates listed that are known to cause a crash in
+        11.7. The call is then put on hold hold (which is the trigger
+        for the crash) and taken off hold. If both the hold and unhold
+        events are received and Asterisk doesn't crash, the test is
+        considered successful. Note that the direction of media is not
+        considered in this test.
+
+properties:
+    minversion: '11.8.0'
+    dependencies:
+        - sipp :
+            version : 'v3.0'
+        - asterisk : 'chan_sip'
+    tags:
+        - SIP
+        - ICE

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Modified: asterisk/trunk/tests/channels/SIP/tests.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/tests.yaml?view=diff&rev=4801&r1=4800&r2=4801
==============================================================================
--- asterisk/trunk/tests/channels/SIP/tests.yaml (original)
+++ asterisk/trunk/tests/channels/SIP/tests.yaml Fri Mar  7 09:58:12 2014
@@ -43,6 +43,7 @@
     - test: 'sip_hold'
     - test: 'sip_hold_direct_media'
     - test: 'sip_hold_no_moh'
+    - test: 'sip_hold_ice'
     - test: 'header_parsing'
     - test: 'use_contact_from_200'
     - test: 'generic_ccss'

Added: asterisk/trunk/tests/channels/pjsip/hold_ice/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/hold_ice/configs/ast1/extensions.conf?view=auto&rev=4801
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/hold_ice/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/hold_ice/configs/ast1/extensions.conf Fri Mar  7 09:58:12 2014
@@ -1,0 +1,10 @@
+[general]
+PHONE_TO_DIAL=PJSIP/phone_B
+
+[default]
+; Dial with no options; use bridge set up based on peer definitions
+exten => devicehint,hint,PJSIP/phone_A
+
+exten => basicdial,1,NoOp()
+	same => n,Dial(PJSIP/phone_B,,g)
+	same => n,Hangup()

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Added: asterisk/trunk/tests/channels/pjsip/hold_ice/configs/ast1/pjsip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/hold_ice/configs/ast1/pjsip.conf?view=auto&rev=4801
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/hold_ice/configs/ast1/pjsip.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/hold_ice/configs/ast1/pjsip.conf Fri Mar  7 09:58:12 2014
@@ -1,0 +1,30 @@
+[local]
+type=transport
+protocol=udp
+bind=0.0.0.0
+
+[phone_A]
+type=aor
+contact=sip:phone_A at 127.0.0.2:6080
+
+[phone_A]
+type=endpoint
+aors=phone_A
+context=default
+disallow=all
+allow=ulaw
+direct_media=no
+ice_support=yes
+
+[phone_B]
+type=aor
+contact=sip:phone_B at 127.0.0.3:6081
+
+[phone_B]
+type=endpoint
+aors=phone_B
+context=default
+disallow=all
+allow=ulaw
+direct_media=no
+ice_support=yes

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Added: asterisk/trunk/tests/channels/pjsip/hold_ice/configs/ast1/rtp.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/hold_ice/configs/ast1/rtp.conf?view=auto&rev=4801
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/hold_ice/configs/ast1/rtp.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/hold_ice/configs/ast1/rtp.conf Fri Mar  7 09:58:12 2014
@@ -1,0 +1,5 @@
+[general]
+rtpstart=10000
+rtpend=20000
+icesupport=yes
+stunaddr=stun.l.google.com:19302

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Added: asterisk/trunk/tests/channels/pjsip/hold_ice/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/hold_ice/run-test?view=auto&rev=4801
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/hold_ice/run-test (added)
+++ asterisk/trunk/tests/channels/pjsip/hold_ice/run-test Fri Mar  7 09:58:12 2014
@@ -1,0 +1,154 @@
+#!/usr/bin/env python
+"""
+Copyright (C) 2014, Digium, Inc.
+Jonathan Rose <jrose at digium.com>
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+"""
+
+import sys
+import logging
+
+sys.path.append("lib/python")
+
+from asterisk.test_case import TestCase
+from asterisk.sipp import SIPpScenario
+from twisted.internet import reactor
+
+LOGGER = logging.getLogger(__name__)
+INJECT_FILE = "inject.csv"
+
+
+class SIPHold(TestCase):
+    """TestCase to execute and evaluate PJSIP hold/unhold with ICE scenario """
+
+    def __init__(self):
+        """ Constructor """
+        super(SIPHold, self).__init__()
+        self.create_asterisk()
+
+        self.sipp_phone_a_scenario = {'scenario': 'phone_A.xml',
+                                      '-i': '127.0.0.2',
+                                      '-p': ' 6080',
+                                      '-inf': INJECT_FILE}
+        self.sipp_phone_b_scenario = {'scenario': 'phone_B.xml',
+                                      '-i': '127.0.0.3',
+                                      '-p': '6081',
+                                      '-inf': INJECT_FILE}
+
+        self.moh_start_event = False
+        self.moh_stop_event = False
+        self.check_list_success_events = 0
+
+    def ami_connect(self, ami):
+        """ Reaction to new AMI connection
+
+        :param ami: AMI connection that was established
+        """
+        super(SIPHold, self).ami_connect(ami)
+
+        ami.registerEvent('TestEvent', self.test_event_handler)
+
+        ami.registerEvent('MusicOnHoldStart', self.moh_start_event_handler)
+        ami.registerEvent('MusicOnHoldStop', self.moh_stop_event_handler)
+
+        LOGGER.info("Starting SIP scenario")
+        self.execute_scenario()
+
+    def execute_scenario(self):
+        """Execute sipp scenarios and check results for a single test phase
+        """
+        sipp_a = SIPpScenario(self.test_name, self.sipp_phone_a_scenario)
+        sipp_b = SIPpScenario(self.test_name, self.sipp_phone_b_scenario)
+
+        # Start up the listener first - Phone A calls Phone B
+        sipp_b.run(self)
+        sipp_a.run(self)
+
+    def handle_checklist_create(self, ami, event):
+        """ Reacts to ICECHECKLISTCREATE user events
+
+        :param ami AMI connection the event was received from
+        :param event Event that was received
+        """
+        if event.get('result') == 'SUCCESS':
+            self.check_list_success_events += 1
+            LOGGER.debug("Received create check list success")
+            if self.check_list_success_events == 2 and \
+                    self.moh_stop_event:
+                self.evaluate_success()
+                self.stop_reactor()
+        else:
+            LOGGER.error("Failed to create check list - test failed")
+            self.set_passed(False)
+            self.stop_reactor()
+
+    def test_event_handler(self, ami, event):
+        """ Reacts to User Events and routes them based on their State Value
+
+        :param ami AMI connection the event was received from
+        :param event Event that was received
+        """
+        if event.get('state') == 'ICECHECKLISTCREATE':
+            self.handle_checklist_create(ami, event)
+
+    def moh_start_event_handler(self, ami, event):
+        """ Reacts to music on hold start events and tallies them.
+
+        :param ami: AMI connection the event was received from
+        :param event: Event that was received
+        """
+        LOGGER.debug("Received MOH start event")
+        self.moh_start_event = True
+
+    def moh_stop_event_handler(self, ami, event):
+        """ Reacts to music on hold stop events and tallies them.
+
+        :param ami: AMI connection the event was received from
+        :param event: Event that was received
+        """
+
+        LOGGER.debug("Received MOH stop event")
+        self.moh_stop_event = True
+        if (self.check_list_success_events == 2):
+            self.evaluate_success()
+            self.stop_reactor()
+
+    def run(self):
+        """ Run the test and create an AMI connection """
+
+        super(SIPHold, self).run()
+        self.create_ami_factory()
+
+    def evaluate_success(self):
+        """ Determine whether all of the conditions for passing the test
+            have been met and raise passed flag accordingly.
+        """
+        if not self.moh_start_event:
+            LOGGER.error("Failed to receive MOH start event")
+            self.set_passed(False)
+        if not self.moh_stop_event:
+            LOGGER.error("Failed to receive MOH stop event")
+            self.set_passed(False)
+        if not self.check_list_success_events == 2:
+            LOGGER.error("Expected 2 check list creation events. Got %d\n" %
+                         self.check_list_success_events)
+            self.set_passed(False)
+
+        self.set_passed(True)
+
+
+def main():
+    test = SIPHold()
+    reactor.run()
+    if test.passed:
+        return 0
+    return 1
+
+
+if __name__ == "__main__":
+    sys.exit(main())
+
+
+# vim:sw=4:ts=4:expandtab:textwidth=79

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