[Asterisk-code-review] testsuite: add test to verify reinvites with rewrite contact (testsuite[13])
Torrey Searle
asteriskteam at digium.com
Wed Oct 24 07:47:23 CDT 2018
Torrey Searle has uploaded this change for review. ( https://gerrit.asterisk.org/10532
Change subject: testsuite: add test to verify reinvites with rewrite_contact
......................................................................
testsuite: add test to verify reinvites with rewrite_contact
Verify that contact does not get modified if routset exists but
the reinvite does not contain Record-Route headers
ASTERISK-28129 #close
Change-Id: Ib6a2cbb21acac4837131f5d048b148658160367e
---
A tests/channels/pjsip/nat/rewrite_contact/route_set_reinvite/configs/ast1/extensions.conf
A tests/channels/pjsip/nat/rewrite_contact/route_set_reinvite/configs/ast1/pjsip.conf
A tests/channels/pjsip/nat/rewrite_contact/route_set_reinvite/sipp/uac-route-set.xml
A tests/channels/pjsip/nat/rewrite_contact/route_set_reinvite/test-config.yaml
M tests/channels/pjsip/nat/rewrite_contact/tests.yaml
5 files changed, 207 insertions(+), 0 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/32/10532/1
diff --git a/tests/channels/pjsip/nat/rewrite_contact/route_set_reinvite/configs/ast1/extensions.conf b/tests/channels/pjsip/nat/rewrite_contact/route_set_reinvite/configs/ast1/extensions.conf
new file mode 100644
index 0000000..7b75b3e
--- /dev/null
+++ b/tests/channels/pjsip/nat/rewrite_contact/route_set_reinvite/configs/ast1/extensions.conf
@@ -0,0 +1,5 @@
+[default]
+exten => echo,1,NoOp()
+same => n,Answer()
+same => n,Echo()
+same => n,Hangup()
diff --git a/tests/channels/pjsip/nat/rewrite_contact/route_set_reinvite/configs/ast1/pjsip.conf b/tests/channels/pjsip/nat/rewrite_contact/route_set_reinvite/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..f2ec207
--- /dev/null
+++ b/tests/channels/pjsip/nat/rewrite_contact/route_set_reinvite/configs/ast1/pjsip.conf
@@ -0,0 +1,14 @@
+[system]
+type=system
+timer_t1=100
+timer_b=6400
+
+[local]
+type = transport
+bind = 127.0.0.1:5060
+
+[sipp]
+type = endpoint
+context = default
+allow = ulaw
+rewrite_contact = yes
diff --git a/tests/channels/pjsip/nat/rewrite_contact/route_set_reinvite/sipp/uac-route-set.xml b/tests/channels/pjsip/nat/rewrite_contact/route_set_reinvite/sipp/uac-route-set.xml
new file mode 100644
index 0000000..0908578
--- /dev/null
+++ b/tests/channels/pjsip/nat/rewrite_contact/route_set_reinvite/sipp/uac-route-set.xml
@@ -0,0 +1,149 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic Sipstone UAC">
+ <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]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Record-Route: <sip:1.2.3.4:5061;lr>
+ Record-Route: <sip:127.0.0.1:5062;lr>
+ Contact: <sip:1.2.3.4:5063;transport=[transport]>
+ 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:0 PCMU/8000
+
+ ]]>
+ </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]:[remote_port]>[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 ="10000" />
+
+ <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]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 INVITE
+ Contact: <sip:1.2.3.4:5063;transport=[transport]>
+ 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:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ 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]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 ACK
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="BYE">
+ <action>
+ <ereg regexp="BYE sip:1.2.3.4:5063.*"
+ search_in="msg"
+ check_it="true"
+ assign_to="1"/>
+ <ereg regexp="Route: <sip:127.0.0.1:5061;lr>\r\nRoute: <sip:127.0.0.1:5062;lr>"
+ search_in="msg"
+ check_it="true"
+ assign_to="2"/>
+ </action>
+ </recv>
+
+ <Reference variables="1,2" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+</scenario>
diff --git a/tests/channels/pjsip/nat/rewrite_contact/route_set_reinvite/test-config.yaml b/tests/channels/pjsip/nat/rewrite_contact/route_set_reinvite/test-config.yaml
new file mode 100644
index 0000000..133fbd1
--- /dev/null
+++ b/tests/channels/pjsip/nat/rewrite_contact/route_set_reinvite/test-config.yaml
@@ -0,0 +1,38 @@
+testinfo:
+ summary: 'Ensure that proper URI is rewritten on SIP responses'
+ description: |
+ 'This test has SIPp place a call to Asterisk. The SIPp scenario
+ represents a proxy in the path to some endpoint. The INVITE that the
+ SIPp scenario sends has Record-Route headers in it. We ensure that
+ Asterisk does not attempt to rewrite the Contact header in the INVITE
+ despite the fact that the rewrite_contact option is enabled. We instead
+ ensure that the top-most Record-Route header is rewritten. Also we verify
+ that if a re-invite w/o a Record-Route does not cause the contact to be
+ updated. We then hang up the call, and we ensure that the request URI
+ and the route set in the BYE is correct.'
+
+test-modules:
+ test-object:
+ config-section: sipp-config
+ typename: 'sipp.SIPpAMIActionTestCase'
+
+sipp-config:
+ memcheck-delay-stop: 7
+ fail-on-any: True
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'uac-route-set.xml', '-p': '5061', '-s': 'echo'} }
+ ami-action:
+ delay: 2
+ args:
+ Action: 'Hangup'
+ Channel: '/PJSIP/sipp-.*/'
+
+properties:
+ dependencies:
+ - sipp:
+ version: 'v3.0'
+ - asterisk: 'res_pjsip'
+ tags:
+ - pjsip
diff --git a/tests/channels/pjsip/nat/rewrite_contact/tests.yaml b/tests/channels/pjsip/nat/rewrite_contact/tests.yaml
index 8f1a3e1..9c04329 100644
--- a/tests/channels/pjsip/nat/rewrite_contact/tests.yaml
+++ b/tests/channels/pjsip/nat/rewrite_contact/tests.yaml
@@ -1,3 +1,4 @@
tests:
- test: 'route_set_response'
- test: 'route_set_request'
+ - test: 'route_set_reinvite'
--
To view, visit https://gerrit.asterisk.org/10532
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Gerrit-Project: testsuite
Gerrit-Branch: 13
Gerrit-MessageType: newchange
Gerrit-Change-Id: Ib6a2cbb21acac4837131f5d048b148658160367e
Gerrit-Change-Number: 10532
Gerrit-PatchSet: 1
Gerrit-Owner: Torrey Searle <tsearle at gmail.com>
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