[asterisk-commits] Add a NAPTR test involving restricted transports. (testsuite[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Sat Jun 13 15:50:57 CDT 2015
Matt Jordan has submitted this change and it was merged.
Change subject: Add a NAPTR test involving restricted transports.
......................................................................
Add a NAPTR test involving restricted transports.
Two NAPTR records exist for example.com in this test. The one with the
lowest preference value points to a TCP SRV record. The one with the
highest preference value points to a UDP SRV record. pjsip.conf has no
TCP transport set up. So Asterisk should skip the TCP SRV record lookup
and skip straight to the UDP SRV record lookup.
A SIPp scenario listens at the UDP destination. The test succeeds if the
SIPp scenario succeeds.
ASTERISK-25014 #close
Reported by Matt Jordan
Change-Id: I2940dcf61a5efcf4908a7f934e5c6d8bde5e24fc
---
A tests/channels/pjsip/resolver/naptr/restricted_transport/configs/ast1/extensions.conf
A tests/channels/pjsip/resolver/naptr/restricted_transport/configs/ast1/pjsip.conf
A tests/channels/pjsip/resolver/naptr/restricted_transport/configs/ast1/resolver_unbound.conf
A tests/channels/pjsip/resolver/naptr/restricted_transport/dns_zones/example.com
A tests/channels/pjsip/resolver/naptr/restricted_transport/sipp/uas.xml
A tests/channels/pjsip/resolver/naptr/restricted_transport/test-config.yaml
M tests/channels/pjsip/resolver/naptr/tests.yaml
7 files changed, 172 insertions(+), 0 deletions(-)
Approvals:
Matt Jordan: Looks good to me, approved; Verified
Joshua Colp: Looks good to me, but someone else must approve
diff --git a/tests/channels/pjsip/resolver/naptr/restricted_transport/configs/ast1/extensions.conf b/tests/channels/pjsip/resolver/naptr/restricted_transport/configs/ast1/extensions.conf
new file mode 100644
index 0000000..4a2d2ba
--- /dev/null
+++ b/tests/channels/pjsip/resolver/naptr/restricted_transport/configs/ast1/extensions.conf
@@ -0,0 +1,6 @@
+[default]
+
+exten => s,1,NoOp()
+ same => n,Wait(1)
+ same => n,Dial(PJSIP/rick/sip:example.com)
+ same => n,Hangup()
diff --git a/tests/channels/pjsip/resolver/naptr/restricted_transport/configs/ast1/pjsip.conf b/tests/channels/pjsip/resolver/naptr/restricted_transport/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..2b61a19
--- /dev/null
+++ b/tests/channels/pjsip/resolver/naptr/restricted_transport/configs/ast1/pjsip.conf
@@ -0,0 +1,9 @@
+[transport=udp]
+type=transport
+protocol=udp
+bind=0.0.0.0:5060
+
+[rick]
+type=endpoint
+context=default
+allow=!all,ulaw,alaw,g722
diff --git a/tests/channels/pjsip/resolver/naptr/restricted_transport/configs/ast1/resolver_unbound.conf b/tests/channels/pjsip/resolver/naptr/restricted_transport/configs/ast1/resolver_unbound.conf
new file mode 100644
index 0000000..38ef153
--- /dev/null
+++ b/tests/channels/pjsip/resolver/naptr/restricted_transport/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/naptr/restricted_transport/dns_zones/example.com b/tests/channels/pjsip/resolver/naptr/restricted_transport/dns_zones/example.com
new file mode 100644
index 0000000..7e563df
--- /dev/null
+++ b/tests/channels/pjsip/resolver/naptr/restricted_transport/dns_zones/example.com
@@ -0,0 +1,33 @@
+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"
+ ),
+
+ NAPTR('example.com', 50, 50, 'S', 'SIP+D2T', '', '_sip._tcp.example.com'),
+ NAPTR('example.com', 50, 90, 'S', 'SIP+D2U', '', '_sip._udp.example.com'),
+ SRV('_sip._tcp.example.com', 0, 1, 5061, 'pbx.example.com'),
+ SRV('_sip._udp.example.com', 0, 1, 5061, 'pbx.example.com'),
+ A('pbx.example.com', '127.0.0.1'),
+]
diff --git a/tests/channels/pjsip/resolver/naptr/restricted_transport/sipp/uas.xml b/tests/channels/pjsip/resolver/naptr/restricted_transport/sipp/uas.xml
new file mode 100644
index 0000000..e72519e
--- /dev/null
+++ b/tests/channels/pjsip/resolver/naptr/restricted_transport/sipp/uas.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/naptr/restricted_transport/test-config.yaml b/tests/channels/pjsip/resolver/naptr/restricted_transport/test-config.yaml
new file mode 100644
index 0000000..c2483fc
--- /dev/null
+++ b/tests/channels/pjsip/resolver/naptr/restricted_transport/test-config.yaml
@@ -0,0 +1,55 @@
+testinfo:
+ summary: 'Test NAPTR record order to ensure lowest order is used.'
+ description: |
+ 'This test verifies that when a NAPTR record results in an unsupported
+ transport, that Asterisk fails over to the NAPTR record with the next
+ highest preference value.
+
+ A call is placed to example.com without a transport specified resulting
+ in a NAPTR lookup. The lowest preference record points to an SRV entry
+ for a TCP transport. Asterisk should skip this SRV record since TCP is
+ not supported. The next lowest preference NAPTR record points to an SRV
+ entry for UDP. A SIPp instance is listening on UDP, so the test will
+ succeed if UDP is used.'
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+ modules:
+ -
+ config-section: originator
+ typename: 'pluggable_modules.Originator'
+ -
+ config-section: dns-server-config
+ typename: 'dns_server.DNSServer'
+
+test-object-config:
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'uas.xml', '-i': '127.0.0.1',
+ '-p': '5061'} }
+
+originator:
+ trigger: 'ami_connect'
+ ignore-originate-failure: 'no'
+ id: '0'
+ channel: 'Local/s 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/naptr/tests.yaml b/tests/channels/pjsip/resolver/naptr/tests.yaml
index e19e6f1..469a3d0 100644
--- a/tests/channels/pjsip/resolver/naptr/tests.yaml
+++ b/tests/channels/pjsip/resolver/naptr/tests.yaml
@@ -6,3 +6,4 @@
- test: 'order'
- test: 'preference'
- test: 'preference_failover'
+ - test: 'restricted_transport'
--
To view, visit https://gerrit.asterisk.org/637
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: I2940dcf61a5efcf4908a7f934e5c6d8bde5e24fc
Gerrit-PatchSet: 3
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>
More information about the asterisk-commits
mailing list