[asterisk-dev] Change in testsuite[master]: pjsip: Add test for OPTIONS requests received in-dialog.
Matt Jordan (Code Review)
asteriskteam at digium.com
Fri Apr 10 08:27:06 CDT 2015
Matt Jordan has submitted this change and it was merged.
Change subject: pjsip: Add test for OPTIONS requests received in-dialog.
......................................................................
pjsip: Add test for OPTIONS requests received in-dialog.
This test establishes a session with Asterisk. Once established an
OPTIONS request is sent in-dialog. Asterisk is expected to respond
with a 200 OK. If this does not occur the test fails.
ASTERISK-24862 #close
Reported by: yaron nahum
Change-Id: I137c66e4faad5212040f6aded6be2aac20fc473d
---
A tests/channels/pjsip/in_dialog_options/configs/ast1/extensions.conf
A tests/channels/pjsip/in_dialog_options/configs/ast1/pjsip.conf
A tests/channels/pjsip/in_dialog_options/sipp/invite_with_options.xml
A tests/channels/pjsip/in_dialog_options/test-config.yaml
M tests/channels/pjsip/tests.yaml
5 files changed, 168 insertions(+), 0 deletions(-)
Approvals:
Matt Jordan: Looks good to me, approved; Verified
diff --git a/tests/channels/pjsip/in_dialog_options/configs/ast1/extensions.conf b/tests/channels/pjsip/in_dialog_options/configs/ast1/extensions.conf
new file mode 100644
index 0000000..711c6cb
--- /dev/null
+++ b/tests/channels/pjsip/in_dialog_options/configs/ast1/extensions.conf
@@ -0,0 +1,4 @@
+[default]
+exten => playback,1,Answer()
+same => n,Playback(demo-congrats)
+same => n,Hangup()
diff --git a/tests/channels/pjsip/in_dialog_options/configs/ast1/pjsip.conf b/tests/channels/pjsip/in_dialog_options/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..79da0da
--- /dev/null
+++ b/tests/channels/pjsip/in_dialog_options/configs/ast1/pjsip.conf
@@ -0,0 +1,15 @@
+[local-transport-template](!)
+type=transport
+bind=127.0.0.1
+
+[local-transport-udp](local-transport-template)
+protocol=udp
+
+[endpoint-template-ipv4](!)
+type=endpoint
+context=default
+allow=!all,ulaw,alaw
+media_address=127.0.0.1
+
+[call](endpoint-template-ipv4)
+transport=local-transport-udp
diff --git a/tests/channels/pjsip/in_dialog_options/sipp/invite_with_options.xml b/tests/channels/pjsip/in_dialog_options/sipp/invite_with_options.xml
new file mode 100644
index 0000000..18c3a8a
--- /dev/null
+++ b/tests/channels/pjsip/in_dialog_options/sipp/invite_with_options.xml
@@ -0,0 +1,120 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to Asterisk and once answered send an OPTIONS request in-dialog">
+ <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:[service]@[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="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:[service]@[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[
+
+ OPTIONS sip:playback@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 OPTIONS
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Length: [len]
+ ]]>
+ </send>
+
+ <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:[service]@[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>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE 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: 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>
+
diff --git a/tests/channels/pjsip/in_dialog_options/test-config.yaml b/tests/channels/pjsip/in_dialog_options/test-config.yaml
new file mode 100644
index 0000000..4e5692d
--- /dev/null
+++ b/tests/channels/pjsip/in_dialog_options/test-config.yaml
@@ -0,0 +1,27 @@
+testinfo:
+ summary: 'Tests handling of an in-dialog OPTIONS'
+ description: |
+ This test runs a SIPp scenario which establishes a session with Asterisk.
+ Once the session has been established an OPTIONS request is sent in-dialog.
+ Asterisk is expected to respond with a 200 OK. If this does not occur then
+ the test fails.
+
+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': 'invite_with_options.xml', '-i': '127.0.0.1', '-p': '5061', '-d': '5000', '-s': 'call'} }
+
+properties:
+ minversion: '13.4.0'
+ dependencies:
+ - app : 'sipp'
+ - asterisk : 'res_pjsip'
+ tags:
+ - pjsip
diff --git a/tests/channels/pjsip/tests.yaml b/tests/channels/pjsip/tests.yaml
index 54b9d35..ddc655a 100644
--- a/tests/channels/pjsip/tests.yaml
+++ b/tests/channels/pjsip/tests.yaml
@@ -35,3 +35,5 @@
- test: 'keep_alive'
- test: 'endpoint_identify'
- test: 'rpid_immediate'
+ - test: 'in_dialog_options'
+
--
To view, visit https://gerrit.asterisk.org/37
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: I137c66e4faad5212040f6aded6be2aac20fc473d
Gerrit-PatchSet: 1
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>
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