[asterisk-commits] mmichelson: branch group/pimp_my_sip r3655 - in /asterisk/team/group/pimp_my_...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Feb 28 17:04:21 CST 2013


Author: mmichelson
Date: Thu Feb 28 17:04:17 2013
New Revision: 3655

URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=3655
Log:
And add further missing tests.


Added:
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/extensions.conf   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/modules.conf   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/res_sip.conf   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/echo_with_deferred_sdp.xml   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/echo_with_initial_sdp.xml   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/playback_with_deferred_sdp.xml   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/playback_with_initial_sdp.xml   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/test-config.yaml   (with props)

Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/extensions.conf?view=auto&rev=3655
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/extensions.conf (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/extensions.conf Thu Feb 28 17:04:17 2013
@@ -1,0 +1,16 @@
+[default]
+exten => echo,1,Answer()
+same  =>      n,Echo()
+same  =>      n,Hangup()
+
+exten => playback,1,Answer()
+same  =>          n,Playback(hello-world)
+same  =>          n,Hangup()
+
+exten => early,1,Progress()
+same  =>       n,Playback(hello-world,noanswer)
+same  =>       n,Hangup(INTERWORKING)
+
+;This dialstring can be altered once endpoints can be used directly
+exten => bob,1,Dial(Gulp/sip:bob at 127.0.0.1:5062)
+same  =>     n,Hangup()

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    svn:keywords = Author Date Id Revision

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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/modules.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/modules.conf?view=auto&rev=3655
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/modules.conf (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/modules.conf Thu Feb 28 17:04:17 2013
@@ -1,0 +1,19 @@
+[modules]
+autoload = no
+preload => res_sorcery_config.so
+preload => res_sorcery_memory.so
+load => res_sip.so
+load => res_sip_logger.so
+load => res_sip_session.so
+load => res_sip_sdp_audio.so
+load => res_sip_endpoint_identifier_ip.so
+load => res_sip_authenticator_digest.so
+load => res_rtp_asterisk.so
+load => app_playback.so
+load => app_echo.so
+load => format_gsm.so
+load => codec_gsm.so
+load => codec_ulaw.so
+load => pbx_config.so
+load => chan_gulp.so
+

Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/modules.conf
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    svn:eol-style = native

Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/modules.conf
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    svn:keywords = Author Date Id Revision

Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/modules.conf
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    svn:mime-type = text/plain

Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/res_sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/res_sip.conf?view=auto&rev=3655
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/res_sip.conf (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/res_sip.conf Thu Feb 28 17:04:17 2013
@@ -1,0 +1,48 @@
+[local-transport-template](!)
+type=transport
+bind=127.0.0.1
+
+[local-transport6-template](!)
+type=transport
+bind=[::1]
+
+[local-transport-udp](local-transport-template)
+protocol=udp
+
+[local-transport-udp6](local-transport6-template)
+protocol=udp
+
+[local-transport-tcp](local-transport-template)
+protocol=tcp
+
+[endpoint-template](!)
+type=endpoint
+context=default
+allow=!all,ulaw,alaw
+
+; alice is the caller
+[alice](endpoint-template)
+; Place alice-specific options here
+host=127.0.0.1:5061
+auth=alice-auth
+
+; bob is the recipient of outbound calls
+[bob](endpoint-template)
+host=127.0.0.1:5062
+; Place bob-specific options here
+
+[auth-template](!)
+type=auth
+password=swordfish
+
+[alice-auth](auth-template)
+username=alice
+; Place alice-specific auth options here
+
+[bob-auth](auth-template)
+username=bob
+; Place bob-specific auth options here
+; Note: in the first iteration of tests on
+; this page, there will never be any bob-specific
+; auth options because we do not respond properly
+; to auth challenges.

Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/res_sip.conf
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    svn:eol-style = native

Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/res_sip.conf
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    svn:keywords = Author Date Id Revision

Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/configs/ast1/res_sip.conf
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    svn:mime-type = text/plain

Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/echo_with_deferred_sdp.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/echo_with_deferred_sdp.xml?view=auto&rev=3655
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/echo_with_deferred_sdp.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/echo_with_deferred_sdp.xml Thu Feb 28 17:04:17 2013
@@ -1,0 +1,122 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to echo with SDP in ACK">
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[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: Test
+      User-Agent: Test
+      Content-Length: 0
+    ]]>
+  </send>
+
+  <recv response="401" auth="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip:echo@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sipp <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 2 INVITE
+      Contact: sip:test@[local_ip]:[local_port]
+      Max-Forwards: 70
+      [authentication username=alice password=swordfish]
+      Subject: Test
+      User-Agent: Test
+      Content-Length: 0
+    ]]>
+  </send>
+
+  <recv response="100"
+        optional="true">
+  </recv>
+
+  <recv response="200" rtd="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip:echo@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 ACK
+      Contact: sip:test@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: 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
+
+    ]]>
+  </send>
+
+  <pause/>
+
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 3 BYE
+      Contact: sip:test@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: 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/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/echo_with_deferred_sdp.xml
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    svn:eol-style = native

Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/echo_with_deferred_sdp.xml
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    svn:keywords = Author Date Id Revision

Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/echo_with_deferred_sdp.xml
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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/echo_with_initial_sdp.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/echo_with_initial_sdp.xml?view=auto&rev=3655
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/echo_with_initial_sdp.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/echo_with_initial_sdp.xml Thu Feb 28 17:04:17 2013
@@ -1,0 +1,132 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to echo with SDP in initial INVITE">
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[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: Test
+      User-Agent: 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
+
+    ]]>
+  </send>
+
+  <recv response="401" auth="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip:echo@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sipp <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 2 INVITE
+      Contact: sip:test@[local_ip]:[local_port]
+      Max-Forwards: 70
+      [authentication username=alice password=swordfish]
+      Subject: Test
+      User-Agent: 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
+
+    ]]>
+  </send>
+  <recv response="100"
+        optional="true">
+  </recv>
+
+  <recv response="200" rtd="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip:echo@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 ACK
+      Contact: sip:test@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <pause/>
+
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 3 BYE
+      Contact: sip:test@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: 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/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/echo_with_initial_sdp.xml
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Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/echo_with_initial_sdp.xml
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    svn:keywords = Author Date Id Revision

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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/playback_with_deferred_sdp.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/playback_with_deferred_sdp.xml?view=auto&rev=3655
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/playback_with_deferred_sdp.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/playback_with_deferred_sdp.xml Thu Feb 28 17:04:17 2013
@@ -1,0 +1,118 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to playback with SDP in ACK">
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:playback@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[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: Test
+      User-Agent: Test
+      Content-Length: 0
+    ]]>
+  </send>
+
+  <recv response="401" auth="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip:playback@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sipp <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:playback@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 2 INVITE
+      Contact: sip:test@[local_ip]:[local_port]
+      Max-Forwards: 70
+      [authentication username=alice password=swordfish]
+      Subject: Test
+      User-Agent: Test
+      Content-Length: 0
+    ]]>
+  </send>
+
+  <recv response="100"
+        optional="true">
+  </recv>
+
+  <recv response="200" rtd="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip:playback@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 ACK
+      Contact: sip:test@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: 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
+
+    ]]>
+  </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>
+
+  <!-- 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/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/playback_with_deferred_sdp.xml
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    svn:eol-style = native

Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/playback_with_deferred_sdp.xml
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    svn:keywords = Author Date Id Revision

Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/playback_with_deferred_sdp.xml
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    svn:mime-type = text/plain

Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/playback_with_initial_sdp.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/playback_with_initial_sdp.xml?view=auto&rev=3655
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/playback_with_initial_sdp.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/playback_with_initial_sdp.xml Thu Feb 28 17:04:17 2013
@@ -1,0 +1,119 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to playback with SDP in initial INVITE">
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:playback@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[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: Test
+      User-Agent: 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
+
+    ]]>
+  </send>
+
+  <recv response="401" auth="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip:playback@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sipp <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:playback@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 2 INVITE
+      Contact: sip:test@[local_ip]:[local_port]
+      Max-Forwards: 70
+      [authentication username=alice password=swordfish]
+      Subject: Test
+      User-Agent: Test
+      Content-Length: 0
+    ]]>
+  </send>
+
+  <recv response="100"
+        optional="true">
+  </recv>
+
+  <recv response="200" rtd="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip:playback@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 ACK
+      Contact: sip:test@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: 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>
+
+  <!-- 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/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/playback_with_initial_sdp.xml
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    svn:eol-style = native

Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/playback_with_initial_sdp.xml
------------------------------------------------------------------------------
    svn:keywords = Author Date Id Revision

Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/sipp/playback_with_initial_sdp.xml
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/test-config.yaml?view=auto&rev=3655
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/test-config.yaml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/test-config.yaml Thu Feb 28 17:04:17 2013
@@ -1,0 +1,59 @@
+testinfo:
+    summary:     'Tests incoming calls without authentication'
+    description: |
+        'Run a SIPp scenario that sends various calls to res_sip, which should not be authenticated'
+
+test-modules:
+    test-object:
+        config-section: test-object-config
+        typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+    fail-on-any: False
+    test-iterations:
+        -
+            scenarios:
+                - { 'key-args': {'scenario': 'playback_with_initial_sdp.xml', '-i': '127.0.0.1', '-p': '5061'} }
+        -
+            scenarios:
+                - { 'key-args': {'scenario': 'echo_with_initial_sdp.xml', '-i': '127.0.0.1', '-p': '5061', '-d': '5000'} }
+        -
+            scenarios:
+                - { 'key-args': {'scenario': 'playback_with_deferred_sdp.xml', '-i': '127.0.0.1', '-p': '5061'} }
+        -
+            scenarios:
+                - { 'key-args': {'scenario': 'echo_with_deferred_sdp.xml', '-i': '127.0.0.1', '-p': '5061', '-d': '5000'} }
+
+#        -
+#            scenarios:
+#                 - { 'key-args': {'scenario': 'playback_with_initial_sdp.xml', '-i': '127.0.0.1', '-p': '5061', '-t': 't1'} }
+#        -
+#            scenarios:
+#                 - { 'key-args': {'scenario': 'echo_with_initial_sdp.xml', '-i': '127.0.0.1', '-p': '5062', '-t': 't1', '-d': '5000'} }
+#        -
+#            scenarios:
+#                 - { 'key-args': {'scenario': 'playback_with_deferred_sdp.xml', '-i': '127.0.0.1', '-p': '5066', '-t': 't1'} }
+#        -
+#            scenarios:
+#                 - { 'key-args': {'scenario': 'echo_with_deferred_sdp.xml', '-i': '127.0.0.1', '-p': '5067', '-t': 't1', '-d': '5000'} }
+# 
+#        -
+#            scenarios:
+#                 - { 'target': '[::1]', 'key-args': {'scenario': 'playback_with_initial_sdp.xml', '-i': '[::1]', '-p': '5061'} }
+#        -
+#            scenarios:
+#                 - { 'target': '[::1]', 'key-args': {'scenario': 'echo_with_initial_sdp.xml', '-i': '[::1]', '-p': '5062'} }
+#        -
+#            scenarios:
+#                 - { 'target': '[::1]', 'key-args': {'scenario': 'playback_with_deferred_sdp.xml', '-i': '[::1]', '-p': '5066'} }
+#        -
+#            scenarios:
+#                 - { 'target': '[::1]', 'key-args': {'scenario': 'echo_with_deferred_sdp.xml', '-i': '[::1]', '-p': '5067'} }
+
+properties:
+    minversion: '12.0.0'
+    dependencies:
+        - sipp :
+            version : 'v3.0'
+    tags:
+        - gulp

Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/test-config.yaml
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    svn:eol-style = native

Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/test-config.yaml
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    svn:keywords = Author Date Id Revision

Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/basic_calls/incoming/nominal/authed/userpass/ident_by_host/test-config.yaml
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