[Asterisk-code-review] pjsip/srtp not loaded: Add tests which cover when res srtp i... (testsuite[master])
Joshua Colp
asteriskteam at digium.com
Fri Nov 11 07:18:13 CST 2016
Joshua Colp has uploaded a new change for review. ( https://gerrit.asterisk.org/4395 )
Change subject: pjsip/srtp_not_loaded: Add tests which cover when res_srtp is not loaded.
......................................................................
pjsip/srtp_not_loaded: Add tests which cover when res_srtp is not loaded.
This change adds tests for res_pjsip SDP negotiation when res_srtp
is not loaded. These tests cover:
Normal non-SRTP call into Asterisk that gets connected.
Optimistic SRTP call into Asterisk that falls back to non-SRTP when connected.
Mandatory SRTP call into Asterisk that is rejected with 488.
ASTERISK-26575
Change-Id: I5a8c648bcc991de8d9d5531329e08bc936e19300
---
A tests/channels/pjsip/srtp_not_loaded/configs/ast1/extensions.conf
A tests/channels/pjsip/srtp_not_loaded/configs/ast1/modules.conf.inc
A tests/channels/pjsip/srtp_not_loaded/configs/ast1/pjsip.conf
A tests/channels/pjsip/srtp_not_loaded/sipp/accept_with_optimistic.xml
A tests/channels/pjsip/srtp_not_loaded/sipp/accept_without_srtp.xml
A tests/channels/pjsip/srtp_not_loaded/sipp/decline_with_required.xml
A tests/channels/pjsip/srtp_not_loaded/test-config.yaml
M tests/channels/pjsip/tests.yaml
8 files changed, 291 insertions(+), 0 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/95/4395/1
diff --git a/tests/channels/pjsip/srtp_not_loaded/configs/ast1/extensions.conf b/tests/channels/pjsip/srtp_not_loaded/configs/ast1/extensions.conf
new file mode 100644
index 0000000..cf3408d
--- /dev/null
+++ b/tests/channels/pjsip/srtp_not_loaded/configs/ast1/extensions.conf
@@ -0,0 +1,4 @@
+[default]
+exten => echo,1,Answer()
+same => n,Echo()
+same => n,Hangup()
diff --git a/tests/channels/pjsip/srtp_not_loaded/configs/ast1/modules.conf.inc b/tests/channels/pjsip/srtp_not_loaded/configs/ast1/modules.conf.inc
new file mode 100644
index 0000000..2af782e
--- /dev/null
+++ b/tests/channels/pjsip/srtp_not_loaded/configs/ast1/modules.conf.inc
@@ -0,0 +1 @@
+noload => res_srtp.so
diff --git a/tests/channels/pjsip/srtp_not_loaded/configs/ast1/pjsip.conf b/tests/channels/pjsip/srtp_not_loaded/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..502ff4a
--- /dev/null
+++ b/tests/channels/pjsip/srtp_not_loaded/configs/ast1/pjsip.conf
@@ -0,0 +1,13 @@
+[local-transport-udp]
+protocol=udp
+type=transport
+bind=127.0.0.1
+
+[endpoint-template](!)
+type=endpoint
+context=default
+allow=!all,ulaw,alaw
+
+[alice](endpoint-template)
+media_encryption_optimistic=yes
+media_encryption=sdes
diff --git a/tests/channels/pjsip/srtp_not_loaded/sipp/accept_with_optimistic.xml b/tests/channels/pjsip/srtp_not_loaded/sipp/accept_with_optimistic.xml
new file mode 100644
index 0000000..86e8de9
--- /dev/null
+++ b/tests/channels/pjsip/srtp_not_loaded/sipp/accept_with_optimistic.xml
@@ -0,0 +1,94 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="SRTP Test">
+ <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="200" rtd="true">
+ <action>
+ <ereg regexp="a=crypto:1" search_in="body" check_it="false" 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>
+
diff --git a/tests/channels/pjsip/srtp_not_loaded/sipp/accept_without_srtp.xml b/tests/channels/pjsip/srtp_not_loaded/sipp/accept_without_srtp.xml
new file mode 100644
index 0000000..72a12f5
--- /dev/null
+++ b/tests/channels/pjsip/srtp_not_loaded/sipp/accept_without_srtp.xml
@@ -0,0 +1,93 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="SRTP Test">
+ <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=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_inverse="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>
+
diff --git a/tests/channels/pjsip/srtp_not_loaded/sipp/decline_with_required.xml b/tests/channels/pjsip/srtp_not_loaded/sipp/decline_with_required.xml
new file mode 100644
index 0000000..514548a
--- /dev/null
+++ b/tests/channels/pjsip/srtp_not_loaded/sipp/decline_with_required.xml
@@ -0,0 +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:0 AES_CM_128_HMAC_SHA1_80 inline:qtoB12WnTe19t8vuhcJVRFmoeHhHyF9tcu/4bAkS
+ 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>
+
diff --git a/tests/channels/pjsip/srtp_not_loaded/test-config.yaml b/tests/channels/pjsip/srtp_not_loaded/test-config.yaml
new file mode 100644
index 0000000..c0cc9cd
--- /dev/null
+++ b/tests/channels/pjsip/srtp_not_loaded/test-config.yaml
@@ -0,0 +1,32 @@
+testinfo:
+ summary: 'Tests SDP negotiation when the res_srtp module is not loaded.'
+ description: |
+ 'Run a SIPp scenario that tests SDP negotiation scenarios including:
+ * scenario with non-SRTP offer that is accepted
+ * scenario with SRTP offer that is rejected
+ * scenario with optimistic SRTP offer that is accepted
+ '
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'accept_without_srtp.xml', '-i': '127.0.0.1', '-p': '5061'} }
+ - { 'key-args': {'scenario': 'decline_with_required.xml', '-i': '127.0.0.1', '-p': '5062'} }
+ - { 'key-args': {'scenario': 'accept_with_optimistic.xml', '-i': '127.0.0.1', '-p': '5063'} }
+
+properties:
+ minversion: '13.1.0'
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ - asterisk: 'app_echo'
+ - asterisk: 'res_pjsip'
+ - asterisk: 'res_srtp'
+ tags:
+ - pjsip
diff --git a/tests/channels/pjsip/tests.yaml b/tests/channels/pjsip/tests.yaml
index 3f9ff10..086a61a 100644
--- a/tests/channels/pjsip/tests.yaml
+++ b/tests/channels/pjsip/tests.yaml
@@ -46,4 +46,5 @@
- test: 'rpid_immediate'
- test: 'set_var'
- test: 'srtp_negotiation'
+ - test: 'srtp_not_loaded'
- test: 'user_eq_phone'
--
To view, visit https://gerrit.asterisk.org/4395
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: newchange
Gerrit-Change-Id: I5a8c648bcc991de8d9d5531329e08bc936e19300
Gerrit-PatchSet: 1
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
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