[asterisk-commits] file: testsuite/asterisk/trunk r3426 - in /asterisk/trunk/tests/channels/SIP:...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Aug 10 10:30:56 CDT 2012


Author: file
Date: Fri Aug 10 10:30:50 2012
New Revision: 3426

URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=3426
Log:
Add tests for SDP attribute passthrough.

Review: https://reviewboard.asterisk.org/r/2029/

Added:
    asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/
    asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/configs/
    asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/configs/ast1/
    asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/configs/ast1/extensions.conf   (with props)
    asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/configs/ast1/sip.conf   (with props)
    asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/run-test   (with props)
    asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/
    asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h263.xml   (with props)
    asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h264.xml   (with props)
    asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_speex.xml   (with props)
    asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263.xml   (with props)
    asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264.xml   (with props)
    asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_speex.xml   (with props)
    asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/test-config.yaml   (with props)
Modified:
    asterisk/trunk/tests/channels/SIP/tests.yaml

Added: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/configs/ast1/extensions.conf?view=auto&rev=3426
==============================================================================
--- asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/configs/ast1/extensions.conf Fri Aug 10 10:30:50 2012
@@ -1,0 +1,4 @@
+[default]
+exten => test-h263,1,Dial(SIP/phoneB-h263)
+exten => test-h264,1,Dial(SIP/phoneB-h264)
+exten => test-speex,1,Dial(SIP/phoneB-speex)

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Added: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/configs/ast1/sip.conf?view=auto&rev=3426
==============================================================================
--- asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/configs/ast1/sip.conf (added)
+++ asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/configs/ast1/sip.conf Fri Aug 10 10:30:50 2012
@@ -1,0 +1,38 @@
+[general]
+udpbindaddr=127.0.0.1
+textsupport=yes
+videosupport=yes
+t38pt_udptl=yes
+allowguest=no
+
+[phoneA]
+type=user
+insecure=invite,port
+disallow=all
+allow=ulaw
+allow=h263
+allow=h264
+allow=speex
+
+[phoneB-h263]
+type=peer
+host=127.0.0.3
+port=5063
+disallow=all
+allow=ulaw
+allow=h263
+
+[phoneB-h264]
+type=peer
+host=127.0.0.3
+port=5064
+disallow=all
+allow=ulaw
+allow=h264
+
+[phoneB-speex]
+type=peer
+host=127.0.0.3
+port=5066
+disallow=all
+allow=speex

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Added: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/run-test?view=auto&rev=3426
==============================================================================
--- asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/run-test (added)
+++ asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/run-test Fri Aug 10 10:30:50 2012
@@ -1,0 +1,105 @@
+#!/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.TestCase 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_h263.xml','-i':'127.0.0.2','-p':'5061'},
+                                       {'scenario':'phone_A_h264.xml','-i':'127.0.0.2','-p':'5062'},
+                                       {'scenario':'phone_A_speex.xml','-i':'127.0.0.2','-p':'5065'}]
+        self.sipp_phone_b_scenarios = [{'scenario':'phone_B_h263.xml','-i':'127.0.0.3','-p':'5063'},
+                                       {'scenario':'phone_B_h264.xml','-i':'127.0.0.3','-p':'5064'},
+                                       {'scenario':'phone_B_speex.xml','-i':'127.0.0.3','-p':'5066'}]
+
+        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

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Added: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h263.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h263.xml?view=auto&rev=3426
==============================================================================
--- asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h263.xml (added)
+++ asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h263.xml Fri Aug 10 10:30:50 2012
@@ -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;F=1;I=1;J=1;T=1;K=1;N=1;PAR=255:255;BPP=65535;HRD=1"
+            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>
+

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Added: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h264.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h264.xml?view=auto&rev=3426
==============================================================================
--- asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h264.xml (added)
+++ asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h264.xml Fri Aug 10 10:30:50 2012
@@ -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 profile-level-id=42801E;max-mbps=48600;packetization-mode=1"
+            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>
+

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Added: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_speex.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_speex.xml?view=auto&rev=3426
==============================================================================
--- asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_speex.xml (added)
+++ asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_speex.xml Fri Aug 10 10:30:50 2012
@@ -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>
+

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Added: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263.xml?view=auto&rev=3426
==============================================================================
--- asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263.xml (added)
+++ asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263.xml Fri Aug 10 10:30:50 2012
@@ -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;F=1;I=1;J=1;T=1;K=1;N=1;PAR=255:255;BPP=65535;HRD=1"
+			      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>

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Added: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264.xml?view=auto&rev=3426
==============================================================================
--- asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264.xml (added)
+++ asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264.xml Fri Aug 10 10:30:50 2012
@@ -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 profile-level-id=42801E;max-mbps=48600;packetization-mode=1"
+			      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>

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Added: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_speex.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_speex.xml?view=auto&rev=3426
==============================================================================
--- asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_speex.xml (added)
+++ asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_speex.xml Fri Aug 10 10:30:50 2012
@@ -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>

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Added: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/test-config.yaml?view=auto&rev=3426
==============================================================================
--- asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/test-config.yaml Fri Aug 10 10:30:50 2012
@@ -1,0 +1,18 @@
+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: '11.0.0'
+    dependencies:
+        - sipp :
+            version : 'v3.0'
+    testconditions:
+        - name: 'threads'
+          ignoredThreads:
+            - 'autoservice_run'
+    tags:
+        - SIP

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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=3426&r1=3425&r2=3426
==============================================================================
--- asterisk/trunk/tests/channels/SIP/tests.yaml (original)
+++ asterisk/trunk/tests/channels/SIP/tests.yaml Fri Aug 10 10:30:50 2012
@@ -36,6 +36,7 @@
     - test: 'realtime_sipregs'
     - test: 'realtime_nosipregs'
     - test: 'SDP_offer_answer'
+    - test: 'SDP_attribute_passthrough'
     - test: 'nat_supertest'
     - test: 'pcap_demo'
     - test: 'sip_hold'




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