[Asterisk-code-review] pjsip: Add resolver test for SRV priority with UDP transport. (testsuite[master])

John Bigelow asteriskteam at digium.com
Thu May 21 12:52:21 CDT 2015


John Bigelow has uploaded a new change for review.

  https://gerrit.asterisk.org/505

Change subject: pjsip: Add resolver test for SRV priority with UDP transport.
......................................................................

pjsip: Add resolver test for SRV priority with UDP transport.

This test does a SRV record lookup resulting in two records each with a
different priority and port for the UDP transport. An IPv4 & IPv6 address (A &
AAAA records) is provided for each SRV record. Port 5061 is used if the highest
priority SRV record (lowest number) is chosen and port 5062 otherwise. If both
IPv4 & IPv6 calls do not reach port 5061 (where SIPp is listening on) then the
test fails.

ASTERISK-25009 #close

Change-Id: I8efdf78580cd327452a22863622521fd1a9385bb
---
A tests/channels/pjsip/resolver/srv/priority/configs/ast1/extensions.conf
A tests/channels/pjsip/resolver/srv/priority/configs/ast1/pjsip.conf
A tests/channels/pjsip/resolver/srv/priority/configs/ast1/resolver_unbound.conf
A tests/channels/pjsip/resolver/srv/priority/dns_zones/example.com
A tests/channels/pjsip/resolver/srv/priority/sipp/uas-ipv4.xml
A tests/channels/pjsip/resolver/srv/priority/sipp/uas-ipv6.xml
A tests/channels/pjsip/resolver/srv/priority/test-config.yaml
M tests/channels/pjsip/resolver/srv/tests.yaml
8 files changed, 272 insertions(+), 0 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/05/505/1

diff --git a/tests/channels/pjsip/resolver/srv/priority/configs/ast1/extensions.conf b/tests/channels/pjsip/resolver/srv/priority/configs/ast1/extensions.conf
new file mode 100644
index 0000000..a2fd472
--- /dev/null
+++ b/tests/channels/pjsip/resolver/srv/priority/configs/ast1/extensions.conf
@@ -0,0 +1,11 @@
+[default]
+
+exten => s-ipv4,1,NoOp()
+ same => n,Wait(1)
+ same => n,Dial(PJSIP/jenny/sip:example.com\;transport=udp)
+ same => n,Hangup()
+
+exten => s-ipv6,1,NoOp()
+ same => n,Wait(1)
+ same => n,Dial(PJSIP/forrest/sip:example.com\;transport=udp)
+ same => n,Hangup()
diff --git a/tests/channels/pjsip/resolver/srv/priority/configs/ast1/pjsip.conf b/tests/channels/pjsip/resolver/srv/priority/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..4f45b4a
--- /dev/null
+++ b/tests/channels/pjsip/resolver/srv/priority/configs/ast1/pjsip.conf
@@ -0,0 +1,23 @@
+[transport-ipv4-udp]
+type=transport
+protocol=udp
+bind=127.0.0.1:5060
+
+[transport-ipv6-udp]
+type=transport
+protocol=udp
+bind=[::1]:5060
+
+[jenny]
+type=endpoint
+transport=transport-ipv4-udp
+from_user=jenny
+context=default
+allow=!all,ulaw,alaw,g722
+
+[forrest]
+type=endpoint
+transport=transport-ipv6-udp
+from_user=forrest
+context=default
+allow=!all,ulaw,alaw,g722
diff --git a/tests/channels/pjsip/resolver/srv/priority/configs/ast1/resolver_unbound.conf b/tests/channels/pjsip/resolver/srv/priority/configs/ast1/resolver_unbound.conf
new file mode 100644
index 0000000..38ef153
--- /dev/null
+++ b/tests/channels/pjsip/resolver/srv/priority/configs/ast1/resolver_unbound.conf
@@ -0,0 +1,3 @@
+[general]
+nameserver = 127.0.0.1 at 10053
+resolv =
diff --git a/tests/channels/pjsip/resolver/srv/priority/dns_zones/example.com b/tests/channels/pjsip/resolver/srv/priority/dns_zones/example.com
new file mode 100644
index 0000000..04e4f3b
--- /dev/null
+++ b/tests/channels/pjsip/resolver/srv/priority/dns_zones/example.com
@@ -0,0 +1,34 @@
+zone = [
+    SOA(
+        # For whom we are the authority
+        'example.com',
+
+        # This nameserver's name
+        mname = "ns1.example.com",
+
+        # Mailbox of individual who handles this
+        rname = "root.example.com",
+
+        # Unique serial identifying this SOA data
+        serial = 2003010601,
+
+        # Time interval before zone should be refreshed
+        refresh = "1H",
+
+        # Interval before failed refresh should be retried
+        retry = "1H",
+
+        # Upper limit on time interval before expiry
+        expire = "1H",
+
+        # Minimum TTL
+        minimum = "1H"
+    ),
+
+    SRV('_sip._udp.example.com', 0, 1, 5061, 'main.example.com'),
+    SRV('_sip._udp.example.com', 1, 1, 5062, 'backup.example.com'),
+    A('main.example.com', '127.0.0.1'),
+    A('backup.example.com', '127.0.0.1'),
+    AAAA('main.example.com', '::1'),
+    AAAA('backup.example.com', '::1'),
+]
diff --git a/tests/channels/pjsip/resolver/srv/priority/sipp/uas-ipv4.xml b/tests/channels/pjsip/resolver/srv/priority/sipp/uas-ipv4.xml
new file mode 100644
index 0000000..e72519e
--- /dev/null
+++ b/tests/channels/pjsip/resolver/srv/priority/sipp/uas-ipv4.xml
@@ -0,0 +1,65 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Receive INVITE with audio, immediately answer, and then hangup">
+  <Global variables="remote_tag" />
+  <recv request="INVITE" crlf="true">
+      <action>
+          <!-- Save the from tag. We'll need it when we send our BYE -->
+          <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=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=- 1324901698 1324901698 IN IP4 [local_ip]
+      s=-
+      c=IN IP4 [local_ip]
+      t=0 0
+      m=audio 2226 RTP/AVP 0 101
+      a=sendrecv
+      a=rtpmap:0 PCMU/8000
+      a=rtpmap:101 telephone-event/8000
+
+    ]]>
+  </send>
+
+  <recv request="ACK" rtd="true" crlf="true">
+  </recv>
+
+  <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: ua1 <sip:ua1@[local_ip]:[local_port]>;tag=[call_number]
+      To: [$remote_tag]
+      [last_Call-ID:]
+      CSeq: [cseq] BYE
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200">
+  </recv>
+
+</scenario>
diff --git a/tests/channels/pjsip/resolver/srv/priority/sipp/uas-ipv6.xml b/tests/channels/pjsip/resolver/srv/priority/sipp/uas-ipv6.xml
new file mode 100644
index 0000000..e72519e
--- /dev/null
+++ b/tests/channels/pjsip/resolver/srv/priority/sipp/uas-ipv6.xml
@@ -0,0 +1,65 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Receive INVITE with audio, immediately answer, and then hangup">
+  <Global variables="remote_tag" />
+  <recv request="INVITE" crlf="true">
+      <action>
+          <!-- Save the from tag. We'll need it when we send our BYE -->
+          <ereg regexp=".*(;tag=.*)"
+              header="From:"
+              search_in="hdr"
+              check_it="true"
+              assign_to="remote_tag"/>
+      </action>
+  </recv>
+
+  <send retrans="500">
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=- 1324901698 1324901698 IN IP4 [local_ip]
+      s=-
+      c=IN IP4 [local_ip]
+      t=0 0
+      m=audio 2226 RTP/AVP 0 101
+      a=sendrecv
+      a=rtpmap:0 PCMU/8000
+      a=rtpmap:101 telephone-event/8000
+
+    ]]>
+  </send>
+
+  <recv request="ACK" rtd="true" crlf="true">
+  </recv>
+
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: ua1 <sip:ua1@[local_ip]:[local_port]>;tag=[call_number]
+      To: [$remote_tag]
+      [last_Call-ID:]
+      CSeq: [cseq] BYE
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200">
+  </recv>
+
+</scenario>
diff --git a/tests/channels/pjsip/resolver/srv/priority/test-config.yaml b/tests/channels/pjsip/resolver/srv/priority/test-config.yaml
new file mode 100644
index 0000000..2058f8d
--- /dev/null
+++ b/tests/channels/pjsip/resolver/srv/priority/test-config.yaml
@@ -0,0 +1,70 @@
+testinfo:
+    summary: 'Test SRV priority using a UDP transport'
+    description: |
+        'This test verifies that an SRV record lookup resulting in multiple
+        results with different priorities will use the record with the highest
+        priority (lowest number). 
+
+        A call is placed to example.com over IPv4 & IPv6 resulting in a SRV
+        lookup. If the highest priority SRV record is used for both calls then
+        the IP address of main.example.com and port 5061 is used to reach the
+        SIPp instances. Otherwise the IP address of backup.example.com and port
+        5062 is used and the test will fail to due a SIPp instance not
+        receiving a call.'
+
+test-modules:
+    test-object:
+        config-section: test-object-config
+        typename: 'sipp.SIPpTestCase'
+    modules:
+        -
+            config-section: originator-ipv4
+            typename: 'pluggable_modules.Originator'
+        -
+            config-section: originator-ipv6
+            typename: 'pluggable_modules.Originator'
+        -
+            config-section: dns-server-config
+            typename: 'dns_server.DNSServer'
+
+test-object-config:
+    test-iterations:
+        -
+            scenarios:
+                - { 'key-args': {'scenario': 'uas-ipv4.xml', '-i': '127.0.0.1',
+                                 '-p': '5061'} }
+                - { 'target': '[::1]', 'key-args': {'scenario': 'uas-ipv6.xml',
+                                                    '-i': '[::1]', '-p': '5061'} }
+
+originator-ipv4:
+    trigger: 'scenario_start'
+    scenario-name: 'uas-ipv4.xml'
+    ignore-originate-failure: 'no'
+    id: '0'
+    channel: 'Local/s-ipv4 at default'
+    application: 'Echo'
+    async: 'True'
+
+originator-ipv6:
+    trigger: 'scenario_start'
+    scenario-name: 'uas-ipv6.xml'
+    ignore-originate-failure: 'no'
+    id: '0'
+    channel: 'Local/s-ipv6 at default'
+    application: 'Echo'
+    async: 'True'
+
+dns-server-config:
+    port: 10053
+    python-zones:
+        -
+            example.com
+
+properties:
+    minversion: '14.0.0'
+    dependencies:
+        - app : 'sipp'
+        - asterisk : 'res_pjsip'
+        - asterisk : 'res_resolver_unbound'
+    tags:
+        - pjsip
diff --git a/tests/channels/pjsip/resolver/srv/tests.yaml b/tests/channels/pjsip/resolver/srv/tests.yaml
index 79fcbb9..06274f0 100644
--- a/tests/channels/pjsip/resolver/srv/tests.yaml
+++ b/tests/channels/pjsip/resolver/srv/tests.yaml
@@ -4,3 +4,4 @@
     - test: 'transport_udp'
     - test: 'transport_unspecified'
     - test: 'failover_udp'
+    - test: 'priority'

-- 
To view, visit https://gerrit.asterisk.org/505
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-MessageType: newchange
Gerrit-Change-Id: I8efdf78580cd327452a22863622521fd1a9385bb
Gerrit-PatchSet: 1
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: John Bigelow <jbigelow at digium.com>



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