[svn-commits] file: testsuite/asterisk/trunk r6371 - in /asterisk/trunk/tests/channels/pjsi...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Sat Jan 31 10:31:57 CST 2015
Author: file
Date: Sat Jan 31 10:31:53 2015
New Revision: 6371
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=6371
Log:
pjsip: Add attribute passthrough tests.
Review: https://reviewboard.asterisk.org/r/4394/
Added:
asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/
asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/
asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/ast1/
asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/ast1/pjsip.conf (with props)
asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/run-test (with props)
asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/
asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_h263.xml (with props)
asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_h264.xml (with props)
asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_speex.xml (with props)
asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_h263.xml (with props)
asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_h264.xml (with props)
asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_speex.xml (with props)
asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/test-config.yaml (with props)
asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/basic/sipp/uac-basic-codecs-with-attributes.xml (with props)
Modified:
asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/basic/test-config.yaml
asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/tests.yaml
Added: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/ast1/extensions.conf?view=auto&rev=6371
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/ast1/extensions.conf Sat Jan 31 10:31:53 2015
@@ -1,0 +1,4 @@
+[default]
+exten => test-h263,1,Dial(PJSIP/phoneB-h263)
+exten => test-h264,1,Dial(PJSIP/phoneB-h264)
+exten => test-speex,1,Dial(PJSIP/phoneB-speex)
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/ast1/extensions.conf
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/ast1/extensions.conf
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/ast1/extensions.conf
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/ast1/pjsip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/ast1/pjsip.conf?view=auto&rev=6371
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/ast1/pjsip.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/ast1/pjsip.conf Sat Jan 31 10:31:53 2015
@@ -1,0 +1,47 @@
+[local-transport-udp]
+type=transport
+bind=127.0.0.1
+protocol=udp
+
+[endpoint-template](!)
+type=endpoint
+context=default
+media_address=127.0.0.1
+
+[phoneA](endpoint-template)
+disallow=all
+allow=ulaw
+allow=h263
+allow=h264
+allow=speex
+
+[phoneB-h263](endpoint-template)
+aors=phoneB-h263
+disallow=all
+allow=ulaw
+allow=h263
+
+[phoneB-h263]
+type=aor
+contact=sip:127.0.0.3:5063
+
+[phoneB-h264](endpoint-template)
+type=endpoint
+aors=phoneB-h264
+disallow=all
+allow=ulaw
+allow=h264
+
+[phoneB-h264]
+type=aor
+contact=sip:127.0.0.3:5064
+
+[phoneB-speex](endpoint-template)
+type=peer
+aors=phoneB-speex
+disallow=all
+allow=speex
+
+[phoneB-speex]
+type=aor
+contact=sip:127.0.0.3:5066
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/ast1/pjsip.conf
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/ast1/pjsip.conf
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/configs/ast1/pjsip.conf
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/run-test?view=auto&rev=6371
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/run-test (added)
+++ asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/run-test Sat Jan 31 10:31:53 2015
@@ -1,0 +1,106 @@
+#!/usr/bin/env python
+'''
+Copyright (C) 2012, Digium, Inc.
+Matthew Jordan <mjordan at digium.com>
+Joshua Colp <jcolp at digium.com>
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+'''
+
+import sys
+import os
+import logging
+
+sys.path.append("lib/python")
+
+from asterisk.asterisk import Asterisk
+from asterisk.version import AsteriskVersion
+from asterisk.test_case import TestCase
+from asterisk.sipp import SIPpScenario
+from twisted.internet import reactor
+
+logger = logging.getLogger(__name__)
+TEST_DIR = os.path.dirname(os.path.realpath(__file__))
+
+class SDPPassthrough(TestCase):
+ def __init__(self):
+ TestCase.__init__(self)
+ self.create_asterisk()
+ self.sipp_phone_a_scenarios = [{'scenario':'phone_A_speex.xml','-i':'127.0.0.2','-p':'5065'},
+ {'scenario':'phone_A_h263.xml','-i':'127.0.0.2','-p':'5061'},
+ {'scenario':'phone_A_h264.xml','-i':'127.0.0.2','-p':'5062'},]
+ self.sipp_phone_b_scenarios = [{'scenario':'phone_B_speex.xml','-i':'127.0.0.3','-p':'5066'},
+ {'scenario':'phone_B_h263.xml','-i':'127.0.0.3','-p':'5063'},
+ {'scenario':'phone_B_h264.xml','-i':'127.0.0.3','-p':'5064'},]
+
+ self.passed = True
+ self.__test_counter = 0
+
+ def ami_connect(self, ami):
+ TestCase.ami_connect(self, ami)
+ logger.info("Starting SIP scenario")
+ self.execute_scenarios()
+
+ def execute_scenarios(self):
+ def __check_scenario_a(result):
+ self.__a_finished = True
+ return result
+
+ def __check_scenario_b(result):
+ self.__b_finished = True
+ return result
+
+ def __execute_next_scenario(result):
+ if self.__a_finished and self.__b_finished:
+ self.__test_counter += 1
+ self.reset_timeout()
+ self.execute_scenarios()
+ return result
+
+ if self.__test_counter == len(self.sipp_phone_a_scenarios):
+ logger.info("All scenarios executed")
+ self.stop_reactor()
+ return
+
+ self.sipp_a = SIPpScenario(TEST_DIR, self.sipp_phone_a_scenarios[self.__test_counter])
+ self.sipp_b = SIPpScenario(TEST_DIR, self.sipp_phone_b_scenarios[self.__test_counter])
+
+ # Start up the listener first - Phone A calls Phone B
+ self.__a_finished = False
+ self.__b_finished = False
+ db = self.sipp_b.run(self)
+ da = self.sipp_a.run(self)
+
+ da.addCallback(__check_scenario_a)
+ da.addCallback(__execute_next_scenario)
+ db.addCallback(__check_scenario_b)
+ db.addCallback(__execute_next_scenario)
+
+ def run(self):
+ TestCase.run(self)
+ self.create_ami_factory()
+
+
+def main():
+ test = SDPPassthrough()
+ test.start_asterisk()
+ reactor.run()
+ test.stop_asterisk()
+
+ if not test.passed:
+ return 1
+
+# if not test.sipp_a.passed:
+# return 1
+
+# if not test.sipp_b.passed:
+# return 1
+
+ return 0
+
+if __name__ == "__main__":
+ sys.exit(main())
+
+
+# vim:sw=4:ts=4:expandtab:textwidth=79
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/run-test
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/run-test
------------------------------------------------------------------------------
svn:executable = *
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/run-test
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/run-test
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_h263.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_h263.xml?view=auto&rev=6371
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_h263.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_h263.xml Sat Jan 31 10:31:53 2015
@@ -1,0 +1,97 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Channel Test">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:test-h263@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ User-Agent: Channel Param Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+ m=video 6002 RTP/AVP 34
+ a=rtpmap:34 H263/90000
+ a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=1;CIF16=1;F=1;I=1;J=1;T=1;K=1;N=1;PAR=255:255;BPP=65535;HRD=1
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ <action>
+ <ereg regexp="m=video [0-9]{1,5} RTP/AVP( [0-9]{1,3})+..*a=rtpmap:34 H263/90000.*a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=1;CIF16=1;VGA=0;F=1;I=1;J=1;T=1;K=1;N=1;BPP=65535;HRD=1;PAR=255:255"
+ search_in="body" check_it="true" assign_to="1"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:test-h263@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:test-h263@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:kartoffelsalat@[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>
+
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_h263.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_h263.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_h263.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_h264.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_h264.xml?view=auto&rev=6371
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_h264.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_h264.xml Sat Jan 31 10:31:53 2015
@@ -1,0 +1,97 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Channel Test">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:test-h264@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ User-Agent: Channel Param Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+ m=video 6002 RTP/AVP 99
+ a=rtpmap:99 H264/90000
+ a=fmtp:99 profile-level-id=42801e;packetization-mode=1;max-mbps=48600
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ <action>
+ <ereg regexp="m=video [0-9]{1,5} RTP/AVP( [0-9]{1,3})+..*a=rtpmap:99 H264/90000.*a=fmtp:99 max-mbps=48600;packetization-mode=1;profile-level-id=42801E"
+ search_in="body" check_it="true" assign_to="1"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:test-h264@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:test-h264@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:kartoffelsalat@[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>
+
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_h264.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_h264.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_h264.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_speex.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_speex.xml?view=auto&rev=6371
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_speex.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_speex.xml Sat Jan 31 10:31:53 2015
@@ -1,0 +1,96 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Channel Test">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:test-speex@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ User-Agent: Channel Param Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0 99
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:99 speex/8000
+ a=fmtp:99 sr=8000,mode=any
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ <action>
+ <ereg regexp="a=fmtp:99 sr=8000,mode=any"
+ search_in="body" check_it="false" assign_to="1"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:test-speex@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:test-speex@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:kartoffelsalat@[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>
+
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_speex.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_speex.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_A_speex.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_h263.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_h263.xml?view=auto&rev=6371
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_h263.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_h263.xml Sat Jan 31 10:31:53 2015
@@ -1,0 +1,89 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone B INVITE with H.263 and answer with H.263">
+ <Global variables="global_call_id"/>
+
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp=".*"
+ header="Call-ID:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="global_call_id"/>
+ <ereg regexp="m=video [0-9]{1,5} RTP/AVP( [0-9]{1,3})+..*a=rtpmap:34 H263/90000.*a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=1;CIF16=1;VGA=0;F=1;I=1;J=1;T=1;K=1;N=1;BPP=65535;HRD=1;PAR=255:255"
+ search_in="body" check_it="true" assign_to="1"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ 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:bob@[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=guest3 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=video 6002 RTP/AVP 34
+ a=rtpmap:34 H263/90000
+ a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=1;CIF16=1;F=1;I=1;J=1;T=1;K=1;N=1;PAR=255:255;BPP=65535;HRD=1
+
+ ]]>
+ </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:bob@[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: 0
+ ]]>
+ </send>
+
+</scenario>
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_h263.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_h263.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_h263.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_h264.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_h264.xml?view=auto&rev=6371
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_h264.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_h264.xml Sat Jan 31 10:31:53 2015
@@ -1,0 +1,89 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone B INVITE with H.264 and answer with H.264">
+ <Global variables="global_call_id"/>
+
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp=".*"
+ header="Call-ID:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="global_call_id"/>
+ <ereg regexp="m=video [0-9]{1,5} RTP/AVP( [0-9]{1,3})+..*a=rtpmap:99 H264/90000.*a=fmtp:99 max-mbps=48600;packetization-mode=1;profile-level-id=42801E"
+ search_in="body" check_it="true" assign_to="1"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ 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:bob@[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=guest3 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=video 6002 RTP/AVP 99
+ a=rtpmap:99 H264/90000
+ a=fmtp:99 profile-level-id=42801e;packetization-mode=1;max-mbps=48600
+
+ ]]>
+ </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:bob@[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: 0
+ ]]>
+ </send>
+
+</scenario>
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_h264.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_h264.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_h264.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_speex.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_speex.xml?view=auto&rev=6371
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_speex.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_speex.xml Sat Jan 31 10:31:53 2015
@@ -1,0 +1,90 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone B INVITE with Speex and no attributes">
+ <Global variables="global_call_id"/>
+
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp=".*"
+ header="Call-ID:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="global_call_id"/>
+ <ereg regexp="a=fmtp:99 sr=8000,mode=any"
+ search_in="body" check_it="false" assign_to="1"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ 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:bob@[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=guest3 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0 99
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:99 speex/8000
+ a=fmtp:99 sr=8000,mode=any
+
+ ]]>
+ </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:bob@[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: 0
+ ]]>
+ </send>
+
+</scenario>
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_speex.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_speex.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/sipp/phone_B_speex.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/test-config.yaml?view=auto&rev=6371
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/test-config.yaml Sat Jan 31 10:31:53 2015
@@ -1,0 +1,17 @@
+testinfo:
+ summary: 'Test SDP codec attribute offer/answer and passthrough'
+ description: |
+ This tests SDP codec attribute offer/answer and passthrough. It ensures that attributes
+ that are offered are answered accordingly and that attributes are passed through to a called
+ party.
+
+properties:
+ minversion: '13.2.0'
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ - asterisk : 'res_pjsip'
+ - asterisk : 'res_pjsip_session'
+ - asterisk : 'chan_pjsip'
+ tags:
+ - pjsip
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/test-config.yaml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/test-config.yaml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/test-config.yaml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/basic/sipp/uac-basic-codecs-with-attributes.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/basic/sipp/uac-basic-codecs-with-attributes.xml?view=auto&rev=6371
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/basic/sipp/uac-basic-codecs-with-attributes.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/basic/sipp/uac-basic-codecs-with-attributes.xml Sat Jan 31 10:31:53 2015
@@ -1,0 +1,106 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic Sipstone UAC">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:answer@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: alice <sip:[service]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Codec Test
+ 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 6000 RTP/AVP 9 0 8 101
+ a=rtpmap:9 G722/8000
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:8 PCMA/8000
+ a=rtpmap:101 telephone-event/8000
+ a=fmtp:101 0-16
+ a=ptime:20
+ a=maxptime:20
+ a=sendrecv
+ m=video 6000 RTP/AVP 99 34
+ a=rtpmap:99 H264/90000
+ a=rtpmap:34 H263/90000
+ a=fmtp:99 sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg==
+ a=sendrecv
+
+ ]]>
+ </send>
+
+ <recv response="100" optional="true">
+ </recv>
+
+ <recv response="181" optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ <action>
+ <ereg regexp="m=audio [0-9]{1,5} RTP/AVP 9 0 8 101+..*"
+ search_in="body" check_it="true" assign_to="1"/>
+ <test assign_to="1" variable="1" compare="equal" value=""/>
+ <ereg regexp="m=video [0-9]{1,5} RTP/AVP 99 34+..*"
+ search_in="body" check_it="true" assign_to="2"/>
+ <test assign_to="2" variable="2" compare="equal" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:answer@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: alice <sip:[service]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:alice@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Codec Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <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>
+
[... 63 lines stripped ...]
More information about the svn-commits
mailing list