[Asterisk-code-review] Add test for SIP BYE with Also header (testsuite[master])
Mark Michelson
asteriskteam at digium.com
Fri Jun 12 12:05:48 CDT 2015
Mark Michelson has uploaded a new change for review.
https://gerrit.asterisk.org/649
Change subject: Add test for SIP BYE with Also header
......................................................................
Add test for SIP BYE with Also header
This test launches ten simultaneous SIPp scenarios. Each scenario places
a call to an extension that calls a local channel. The local channel
then calls the Echo() application.
The SIPp scenarios send a BYE with Also header in order to transfer the
;1 side of the local channel to the Echo() application as well.
This test exists both to ensure that a SIP BYE with Also header
functions as expected as well as to ensure that an observed deadlock
that had previously occurred does not happen any longer.
Change-Id: I687d28a725980fb6ccbc9b8a361d2603baee522c
---
A tests/channels/SIP/sip_bye_also/configs/ast1/extensions.conf
A tests/channels/SIP/sip_bye_also/configs/ast1/sip.conf
A tests/channels/SIP/sip_bye_also/sipp/uac-bye-also.xml
A tests/channels/SIP/sip_bye_also/test-config.yaml
M tests/channels/SIP/tests.yaml
5 files changed, 153 insertions(+), 0 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/49/649/1
diff --git a/tests/channels/SIP/sip_bye_also/configs/ast1/extensions.conf b/tests/channels/SIP/sip_bye_also/configs/ast1/extensions.conf
new file mode 100644
index 0000000..c73557a
--- /dev/null
+++ b/tests/channels/SIP/sip_bye_also/configs/ast1/extensions.conf
@@ -0,0 +1,9 @@
+[default]
+
+exten => echo,1,NoOp()
+same => n,Answer()
+same => n,Echo()
+same => n,Hangup()
+
+exten => 200,1,NoOp()
+same => n,Dial(Local/echo at default/n)
diff --git a/tests/channels/SIP/sip_bye_also/configs/ast1/sip.conf b/tests/channels/SIP/sip_bye_also/configs/ast1/sip.conf
new file mode 100644
index 0000000..6b6004d
--- /dev/null
+++ b/tests/channels/SIP/sip_bye_also/configs/ast1/sip.conf
@@ -0,0 +1,9 @@
+[general]
+bindaddr = 127.0.0.1
+bindport = 5060
+
+[sipp]
+type = peer
+context = default
+host = 127.0.0.1
+port = 5061
diff --git a/tests/channels/SIP/sip_bye_also/sipp/uac-bye-also.xml b/tests/channels/SIP/sip_bye_also/sipp/uac-bye-also.xml
new file mode 100644
index 0000000..2f2ccc2
--- /dev/null
+++ b/tests/channels/SIP/sip_bye_also/sipp/uac-bye-also.xml
@@ -0,0 +1,94 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="SIP BYE also">
+ <!-- In client mode (sipp placing calls), the Call-ID MUST be -->
+ <!-- generated by sipp. To do so, use [call_id] keyword. -->
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 0
+ a=rtpmap:8 PCMA/8000
+ a=rtpmap:0 PCMU/8000
+ a=ptime:20
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="181"
+ 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:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]>[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>
+
+ <pause/>
+
+ <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Also: <sip:echo@[remote_ip]:[remote_port]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true">
+ </recv>
+
+</scenario>
+
diff --git a/tests/channels/SIP/sip_bye_also/test-config.yaml b/tests/channels/SIP/sip_bye_also/test-config.yaml
new file mode 100644
index 0000000..bf441de
--- /dev/null
+++ b/tests/channels/SIP/sip_bye_also/test-config.yaml
@@ -0,0 +1,40 @@
+testinfo:
+ summary: 'Test SIP blind transfer using BYE with Also header'
+ description: |
+ 'This test performs the following test:
+ * A SIPp scenario establishes a call to Asterisk
+ * The incoming call calls a Local channel
+ * The Local channel executes the Echo application
+ * The SIPp scenario sends a BYE with Also header to
+ blind transfer one side of the Local channel to the
+ Echo application
+ The scenario is run ten times in parallel. This establishes
+ that the test works, and it also ensures that a previously
+ observed deadlock in this code path does not occur.'
+
+test-modules:
+ test-object:
+ config-section: sipp-config
+ typename: 'sipp.SIPpAMIActionTestCase'
+
+sipp-config:
+ fail-on-any: True
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'uac-bye-also.xml', '-p': '5061', '-m': '10', '-s': '200'} }
+ ami-action:
+ delay: 1
+ args:
+ Action: 'Hangup'
+ Channel: '/Local.*;1/'
+
+properties:
+ minversion: ''
+ dependencies:
+ - python: 'starpy'
+ - sipp:
+ version: 'v3.0'
+ - asterisk: 'chan_sip'
+ tags:
+ - SIP
diff --git a/tests/channels/SIP/tests.yaml b/tests/channels/SIP/tests.yaml
index 9524a68..9796d00 100644
--- a/tests/channels/SIP/tests.yaml
+++ b/tests/channels/SIP/tests.yaml
@@ -72,3 +72,4 @@
- test: 'invite_retransmit'
- test: 'no_ack_dialog_cleanup'
- test: 'no_reinvite_after_491'
+ - test: 'sip_bye_also'
--
To view, visit https://gerrit.asterisk.org/649
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: newchange
Gerrit-Change-Id: I687d28a725980fb6ccbc9b8a361d2603baee522c
Gerrit-PatchSet: 1
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Mark Michelson <mmichelson at digium.com>
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