[Asterisk-code-review] Add test to ensure route set preservation on SIP ACK. (testsuite[master])

Mark Michelson asteriskteam at digium.com
Wed Jun 24 17:03:25 CDT 2015


Mark Michelson has uploaded a new change for review.

  https://gerrit.asterisk.org/712

Change subject: Add test to ensure route set preservation on SIP ACK.
......................................................................

Add test to ensure route set preservation on SIP ACK.

This test has Asterisk originate a call to a SIPp scenario. The
SIPp scenario sends a response that includes Record-Route headers.
The test ensures that the ACK that Asterisk sends preserves the
route set, despite the fact that rewrite_contact has been enabled.

Change-Id: I4c8613508605c5194b262edc0cb10d53f2b467e2
---
A tests/channels/pjsip/nat/rewrite_contact/route_set_response/configs/ast1/extensions.conf
A tests/channels/pjsip/nat/rewrite_contact/route_set_response/configs/ast1/pjsip.conf
A tests/channels/pjsip/nat/rewrite_contact/route_set_response/sipp/uas-route-set.xml
A tests/channels/pjsip/nat/rewrite_contact/route_set_response/test-config.yaml
A tests/channels/pjsip/nat/rewrite_contact/tests.yaml
A tests/channels/pjsip/nat/tests.yaml
M tests/channels/pjsip/tests.yaml
7 files changed, 157 insertions(+), 1 deletion(-)


  git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/12/712/1

diff --git a/tests/channels/pjsip/nat/rewrite_contact/route_set_response/configs/ast1/extensions.conf b/tests/channels/pjsip/nat/rewrite_contact/route_set_response/configs/ast1/extensions.conf
new file mode 100644
index 0000000..c0f0b19
--- /dev/null
+++ b/tests/channels/pjsip/nat/rewrite_contact/route_set_response/configs/ast1/extensions.conf
@@ -0,0 +1,4 @@
+[default]
+exten => sipp,1,NoOp()
+same => n,Dial(PJSIP/sipp)
+same => n,Hangup()
diff --git a/tests/channels/pjsip/nat/rewrite_contact/route_set_response/configs/ast1/pjsip.conf b/tests/channels/pjsip/nat/rewrite_contact/route_set_response/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..2f6fe6e
--- /dev/null
+++ b/tests/channels/pjsip/nat/rewrite_contact/route_set_response/configs/ast1/pjsip.conf
@@ -0,0 +1,14 @@
+[local]
+type = transport
+bind = 127.0.0.1:5060
+
+[sipp]
+type = endpoint
+context = default
+allow = ulaw
+rewrite_contact = yes
+aors = sipp
+
+[sipp]
+type = aor
+contact = sip:sipp at 127.0.0.1:5061
diff --git a/tests/channels/pjsip/nat/rewrite_contact/route_set_response/sipp/uas-route-set.xml b/tests/channels/pjsip/nat/rewrite_contact/route_set_response/sipp/uas-route-set.xml
new file mode 100644
index 0000000..9a1532e
--- /dev/null
+++ b/tests/channels/pjsip/nat/rewrite_contact/route_set_response/sipp/uas-route-set.xml
@@ -0,0 +1,97 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic UAS responder">
+  <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
+  <!-- are saved and used for following messages sent. Useful to test   -->
+  <!-- against stateful SIP proxies/B2BUAs.                             -->
+  <Global variables="remote_tag" />
+  <recv request="INVITE" crlf="true">
+      <!-- Save the from tag. We'll need it when we send our BYE -->
+      <action>
+          <ereg regexp=".*(;tag=.*)"
+              header="From:"
+              search_in="hdr"
+              check_it="true"
+              assign_to="remote_tag"/>
+	  </action>
+  </recv>
+
+  <send retrans="500">
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag01[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Record-Route: <sip:127.0.0.1:5062;lr>
+      Record-Route: <sip:127.0.0.1:5061;lr>
+      Contact: <sip:127.0.0.1:5063>
+      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 request="ACK"
+        rtd="true"
+        crlf="true">
+        <action>
+            <ereg regexp="ACK sip:127.0.0.1:5063.*"
+                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" />
+
+  <pause/>
+
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:[service]@127.0.0.1:[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]SIPpTag01[call_number]
+      To: [$remote_tag]
+      [last_Call-ID:]
+      CSeq: [cseq] BYE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200">
+  </recv>
+
+  <!-- Keep the call open for a while in case the 200 is lost to be     -->
+  <!-- able to retransmit it if we receive the BYE again.               -->
+  <timewait milliseconds="4000"/>
+
+
+  <!-- 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/nat/rewrite_contact/route_set_response/test-config.yaml b/tests/channels/pjsip/nat/rewrite_contact/route_set_response/test-config.yaml
new file mode 100644
index 0000000..bc2a671
--- /dev/null
+++ b/tests/channels/pjsip/nat/rewrite_contact/route_set_response/test-config.yaml
@@ -0,0 +1,36 @@
+testinfo:
+    summary: 'Ensure that proper URI is rewritten on SIP responses'
+    description: |
+        'This test has Asterisk place an outbound call to a SIPp scenario,
+        which represents a proxy in the path to some endpoint. The 200 OK
+        response that SIPp sends has Record-Route headers in it. We ensure
+        that Asterisk does not attempt to rewrite the Contact header in the
+        SIP response but rather rewrites the bottom-most Record-Route header.'
+
+test-modules:
+    test-object:
+        config-section: sipp-config
+        typename: 'sipp.SIPpAMIActionTestCase'
+
+sipp-config:
+    fail-on-any: True
+    test-iterations:
+        -
+            scenarios:
+                - { 'key-args': {'scenario': 'uas-route-set.xml', '-p': '5061'} }
+    ami-action:
+        delay: 1
+        args:
+            Action: 'Originate'
+            Channel: 'PJSIP/sipp'
+            Application: 'Echo'
+
+
+properties:
+    minversion: '13.5.0'
+    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
new file mode 100644
index 0000000..beef6a2
--- /dev/null
+++ b/tests/channels/pjsip/nat/rewrite_contact/tests.yaml
@@ -0,0 +1,2 @@
+tests:
+    - test: 'route_set_response'
diff --git a/tests/channels/pjsip/nat/tests.yaml b/tests/channels/pjsip/nat/tests.yaml
new file mode 100644
index 0000000..e6bdf77
--- /dev/null
+++ b/tests/channels/pjsip/nat/tests.yaml
@@ -0,0 +1,2 @@
+tests:
+    - dir: 'rewrite_contact'
diff --git a/tests/channels/pjsip/tests.yaml b/tests/channels/pjsip/tests.yaml
index b9af685..6351df0 100644
--- a/tests/channels/pjsip/tests.yaml
+++ b/tests/channels/pjsip/tests.yaml
@@ -38,4 +38,5 @@
     - test: 'in_dialog_options'
     - dir: 'resolver'
     - test: 'forward_loop'
-    - dir: 'configuration'
\ No newline at end of file
+    - dir: 'configuration'
+    - dir: 'nat'

-- 
To view, visit https://gerrit.asterisk.org/712
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Gerrit-MessageType: newchange
Gerrit-Change-Id: I4c8613508605c5194b262edc0cb10d53f2b467e2
Gerrit-PatchSet: 1
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Mark Michelson <mmichelson at digium.com>



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