[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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