[asterisk-commits] res pjsip empty: Add test for reply to empty SIP INFO packets (testsuite[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed May 18 13:46:44 CDT 2016
Anonymous Coward #1000019 has submitted this change and it was merged.
Change subject: res_pjsip_empty: Add test for reply to empty SIP INFO packets
......................................................................
res_pjsip_empty: Add test for reply to empty SIP INFO packets
Add new test that checks after establishing a call via an INVITE,
when an INFO packet is recieved with no 'Content-Type' header,
the emtpy INFO module should respond with a 200 - OK packet.
ASTERISK-24986
Change-Id: I7999582041be32e359940be731972ba9387bb397
---
A tests/channels/pjsip/info_empty/configs/ast1/extensions.conf
A tests/channels/pjsip/info_empty/configs/ast1/pjsip.conf
A tests/channels/pjsip/info_empty/sipp/empty.xml
A tests/channels/pjsip/info_empty/test-config.yaml
M tests/channels/pjsip/tests.yaml
5 files changed, 189 insertions(+), 0 deletions(-)
Approvals:
Kevin Harwell: Looks good to me, but someone else must approve
Mark Michelson: Looks good to me, approved
Anonymous Coward #1000019: Verified
Joshua Colp: Looks good to me, but someone else must approve
diff --git a/tests/channels/pjsip/info_empty/configs/ast1/extensions.conf b/tests/channels/pjsip/info_empty/configs/ast1/extensions.conf
new file mode 100644
index 0000000..7476ff3
--- /dev/null
+++ b/tests/channels/pjsip/info_empty/configs/ast1/extensions.conf
@@ -0,0 +1,5 @@
+[default]
+
+exten => empty,1,NoOp()
+ same => n,Answer()
+ same => n,MusicOnHold()
diff --git a/tests/channels/pjsip/info_empty/configs/ast1/pjsip.conf b/tests/channels/pjsip/info_empty/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..33e61b9
--- /dev/null
+++ b/tests/channels/pjsip/info_empty/configs/ast1/pjsip.conf
@@ -0,0 +1,25 @@
+[local]
+type=transport
+protocol=udp
+bind=0.0.0.0
+
+[endpoint_t](!)
+type=endpoint
+context=default
+transport=local
+direct_media=no
+disallow=all
+allow=ulaw
+dtmf_mode=info
+
+[aor_t](!)
+type=aor
+max_contacts=1
+
+;; test
+
+[test](endpoint_t)
+aors=test
+
+[test](aor_t)
+contact=sip:test at 127.0.0.1:5061
diff --git a/tests/channels/pjsip/info_empty/sipp/empty.xml b/tests/channels/pjsip/info_empty/sipp/empty.xml
new file mode 100644
index 0000000..5fb32ec
--- /dev/null
+++ b/tests/channels/pjsip/info_empty/sipp/empty.xml
@@ -0,0 +1,133 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INFO Test">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:empty@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+ To: empty <sip:empty@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: INFO Test
+ 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:empty@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+ To: empty <sip:empty@[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: INFO Test
+ Content-Type: application/sdp
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INFO sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: empty <sip:empty@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 INFO
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Content-Length: [len]
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true">
+ </recv>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INFO sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: empty <sip:empty@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 3 INFO
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Content-Length: [len]
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true">
+ </recv>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INFO sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: empty <sip:empty@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 17 INFO
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Content-Length: [len]
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true">
+ </recv>
+
+ <pause/>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:empty@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+ To: empty <sip:empty@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 18 BYE
+ Contact: sip:test@[local_ip][local_port]
+ Max-Forwards: 70
+ Subject: INFO Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true">
+ </recv>
+
+</scenario>
diff --git a/tests/channels/pjsip/info_empty/test-config.yaml b/tests/channels/pjsip/info_empty/test-config.yaml
new file mode 100644
index 0000000..8e9f286
--- /dev/null
+++ b/tests/channels/pjsip/info_empty/test-config.yaml
@@ -0,0 +1,25 @@
+testinfo:
+ summary: 'Test that asterisk correctly handles EMTPY INFO in SIP packets when using PJSIP.'
+ description:
+ This test checks that after establishing a call via an INVITE, when
+ an INFO packet is recieved with no 'Content-Type' header, the
+ emtpy INFO module should respond with a 200 - OK packet.
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'empty.xml', '-i': '127.0.0.1', '-p': '5061', '-d': '3000'} }
+
+properties:
+ minversion: '13.10.0'
+ dependencies:
+ - app : 'sipp'
+ - asterisk : 'res_pjsip'
+ tags:
+ - pjsip
diff --git a/tests/channels/pjsip/tests.yaml b/tests/channels/pjsip/tests.yaml
index b44fafb..88a9a06 100644
--- a/tests/channels/pjsip/tests.yaml
+++ b/tests/channels/pjsip/tests.yaml
@@ -37,6 +37,7 @@
- test: 'incoming_call_on_second_transport'
- test: 'incoming_calls_without_auth'
- test: 'info_dtmf'
+ - test: 'info_empty'
- test: 'keep_alive'
- test: 'reinvite_early'
- test: 'reinvite_pending'
--
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Gerrit-MessageType: merged
Gerrit-Change-Id: I7999582041be32e359940be731972ba9387bb397
Gerrit-PatchSet: 3
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: snuffy <snuffy22 at gmail.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: snuffy <snuffy22 at gmail.com>
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