[asterisk-commits] kmoore: branch group/pimp_my_sip r3770 - in /asterisk/team/group/pimp_my_sip/...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu May 9 08:16:35 CDT 2013
Author: kmoore
Date: Thu May 9 08:16:32 2013
New Revision: 3770
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=3770
Log:
Add tests for SRTP negotiation
Add tests for SRTP negotiation in SDP for cases where:
* Call declined
** Encryption enabled
*** Unknown cryptographic suite in offer
*** Missing crypto attribute in offer
*** Encryption not offered (attribute present)
*** Missing key
*** Missing tag
*** Missing suite
*** Lifetime present
** Encryption not enabled
* Call accepted
** Encryption not enabled, crypto attribute present and ignored
** Encryption enabled
*** Nominal, single attribute line
*** Nominal, multiple attribute lines with the first offer being unacceptable
Review: https://reviewboard.asterisk.org/r/2482/
(issue ASTERISK-21416)
Added:
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/configs/
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/configs/ast1/
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/configs/ast1/extensions.conf (with props)
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/configs/ast1/res_sip.conf (with props)
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/accept_disabled_with_attrib.xml (with props)
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/accept_multiple_attrib_first_bad.xml (with props)
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/accept_nominal.xml (with props)
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_enabled.xml (with props)
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_attrib.xml (with props)
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_key.xml (with props)
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_suite.xml (with props)
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_tag.xml (with props)
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_not_enabled.xml (with props)
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_unknown_suite.xml (with props)
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_with_lifetime.xml (with props)
asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/test-config.yaml (with props)
Modified:
asterisk/team/group/pimp_my_sip/tests/channels/gulp/tests.yaml
Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/configs/ast1/extensions.conf?view=auto&rev=3770
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/configs/ast1/extensions.conf (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/configs/ast1/extensions.conf Thu May 9 08:16:32 2013
@@ -1,0 +1,4 @@
+[default]
+exten => echo,1,Answer()
+same => n,Echo()
+same => n,Hangup()
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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/configs/ast1/res_sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/configs/ast1/res_sip.conf?view=auto&rev=3770
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/configs/ast1/res_sip.conf (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/configs/ast1/res_sip.conf Thu May 9 08:16:32 2013
@@ -1,0 +1,16 @@
+[local-transport-udp]
+protocol=udp
+type=transport
+bind=127.0.0.1
+
+[endpoint-template](!)
+type=endpoint
+context=default
+allow=!all,ulaw,alaw
+
+; alice supports SRTP
+[alice](endpoint-template)
+media_encryption=sdes
+
+; bob does not support SRTP
+[bob](endpoint-template)
Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/configs/ast1/res_sip.conf
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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/accept_disabled_with_attrib.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/accept_disabled_with_attrib.xml?view=auto&rev=3770
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/accept_disabled_with_attrib.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/accept_disabled_with_attrib.xml Thu May 9 08:16:32 2013
@@ -1,0 +1,90 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Test no encryption enabled with crypto attribute present">
+ <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:bob@[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=guest1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6002 RTP/AVP 0
+ a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:WbTBosdVUZqEb6Htqhn+m3z7wUh4RJVR8nE15GbN
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" 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: 1 ACK
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <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: 2 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>
+
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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/accept_multiple_attrib_first_bad.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/accept_multiple_attrib_first_bad.xml?view=auto&rev=3770
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/accept_multiple_attrib_first_bad.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/accept_multiple_attrib_first_bad.xml Thu May 9 08:16:32 2013
@@ -1,0 +1,96 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Test nominal SRTP negotiation">
+ <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=guest1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6002 RTP/SAVP 0
+ a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:WbTBosdVUZqEb6Htqhn+m3z7wUh4RJVR8nE15GbN|2^20
+ a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj
+ a=crypto:3 AES_CM_128_HMAC_SHA1_32 inline:WbTBosdVUZqEb6Htqhn+m3z7wUh4RJVR8nE15GbN
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </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=crypto:2" search_in="body" check_it="true" assign_to="1"/>
+ </action>
+ </recv>
+ <Reference variables="1"/>
+
+ <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: 1 ACK
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <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: 2 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>
+
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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/accept_nominal.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/accept_nominal.xml?view=auto&rev=3770
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/accept_nominal.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/accept_nominal.xml Thu May 9 08:16:32 2013
@@ -1,0 +1,94 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Test nominal SRTP negotiation">
+ <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=guest1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6002 RTP/SAVP 0
+ a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:WbTBosdVUZqEb6Htqhn+m3z7wUh4RJVR8nE15GbN
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </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=crypto:1" search_in="body" check_it="true" assign_to="1"/>
+ </action>
+ </recv>
+ <Reference variables="1"/>
+
+ <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: 1 ACK
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <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: 2 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>
+
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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_enabled.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_enabled.xml?view=auto&rev=3770
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_enabled.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_enabled.xml Thu May 9 08:16:32 2013
@@ -1,0 +1,53 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Test call declination on unencrypted offer where SRTP is enabled">
+ <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=guest1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6002 RTP/AVP 0
+ a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:WbTBosdVUZqEb6Htqhn+m3z7wUh4RJVR8nE15GbN
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <recv response="488" rtd="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/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_attrib.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_attrib.xml?view=auto&rev=3770
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_attrib.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_attrib.xml Thu May 9 08:16:32 2013
@@ -1,0 +1,52 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Test call declination on missing SRTP crpto attribute">
+ <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=guest1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6002 RTP/SAVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <recv response="488" rtd="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/srtp_negotiation/sipp/decline_no_attrib.xml
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svn:keywords = Author Date Id Revision
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svn:mime-type = text/plain
Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_key.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_key.xml?view=auto&rev=3770
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_key.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_key.xml Thu May 9 08:16:32 2013
@@ -1,0 +1,53 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Test call declination on use of unknown cryptographic suite">
+ <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=guest1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6002 RTP/SAVP 0
+ a=crypto:1 AES_CM_128_HMAC_SHA1_80
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <recv response="488" rtd="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/srtp_negotiation/sipp/decline_no_key.xml
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svn:eol-style = native
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svn:keywords = Author Date Id Revision
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svn:mime-type = text/plain
Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_suite.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_suite.xml?view=auto&rev=3770
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_suite.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_suite.xml Thu May 9 08:16:32 2013
@@ -1,0 +1,53 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Test call declination on use of unknown cryptographic suite">
+ <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=guest1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6002 RTP/SAVP 0
+ a=crypto:1 inline:WbTBosdVUZqEb6Htqhn+m3z7wUh4RJVR8nE15GbN
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <recv response="488" rtd="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/srtp_negotiation/sipp/decline_no_suite.xml
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svn:keywords = Author Date Id Revision
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svn:mime-type = text/plain
Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_tag.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_tag.xml?view=auto&rev=3770
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_tag.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_tag.xml Thu May 9 08:16:32 2013
@@ -1,0 +1,53 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Test call declination on use of unknown cryptographic suite">
+ <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=guest1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6002 RTP/SAVP 0
+ a=crypto:AES_CM_128_HMAC_SHA1_80 inline:WbTBosdVUZqEb6Htqhn+m3z7wUh4RJVR8nE15GbN
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <recv response="488" rtd="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/srtp_negotiation/sipp/decline_no_tag.xml
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svn:keywords = Author Date Id Revision
Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_no_tag.xml
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svn:mime-type = text/plain
Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_not_enabled.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_not_enabled.xml?view=auto&rev=3770
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_not_enabled.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_not_enabled.xml Thu May 9 08:16:32 2013
@@ -1,0 +1,53 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Test call declination on SRTP offer where it is not enabled">
+ <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:bob@[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=guest1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6002 RTP/SAVP 0
+ a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:WbTBosdVUZqEb6Htqhn+m3z7wUh4RJVR8nE15GbN
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <recv response="488" rtd="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/srtp_negotiation/sipp/decline_not_enabled.xml
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svn:eol-style = native
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svn:keywords = Author Date Id Revision
Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_not_enabled.xml
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svn:mime-type = text/plain
Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_unknown_suite.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_unknown_suite.xml?view=auto&rev=3770
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_unknown_suite.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_unknown_suite.xml Thu May 9 08:16:32 2013
@@ -1,0 +1,53 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Test call declination on use of unknown cryptographic suite">
+ <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=guest1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6002 RTP/SAVP 0
+ a=crypto:1 AES_CM_128_HMAC_SHA1_128 inline:WbTBosdVUZqEb6Htqhn+m3z7wUh4RJVR8nE15GbN
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <recv response="488" rtd="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/srtp_negotiation/sipp/decline_unknown_suite.xml
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svn:eol-style = native
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svn:keywords = Author Date Id Revision
Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_unknown_suite.xml
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svn:mime-type = text/plain
Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_with_lifetime.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_with_lifetime.xml?view=auto&rev=3770
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_with_lifetime.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_with_lifetime.xml Thu May 9 08:16:32 2013
@@ -1,0 +1,53 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Test call declination on use of unknown cryptographic suite">
+ <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=guest1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6002 RTP/SAVP 0
+ a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:WbTBosdVUZqEb6Htqhn+m3z7wUh4RJVR8nE15GbN|2^20
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <recv response="488" rtd="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/srtp_negotiation/sipp/decline_with_lifetime.xml
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svn:eol-style = native
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------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/team/group/pimp_my_sip/tests/channels/gulp/srtp_negotiation/sipp/decline_with_lifetime.xml
[... 78 lines stripped ...]
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