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 <mjor...@digium.com> +Joshua Colp <jc...@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 ...] -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- svn-commits mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/svn-commits
