[asterisk-commits] pjsip: Add tests for 'rtp timeout' and 'rtp timeout hold' en... (testsuite[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jul 28 08:50:29 CDT 2015
Matt Jordan has submitted this change and it was merged.
Change subject: pjsip: Add tests for 'rtp_timeout' and 'rtp_timeout_hold' endpoint options.
......................................................................
pjsip: Add tests for 'rtp_timeout' and 'rtp_timeout_hold' endpoint options.
These tests verify that if the 'rtp_timeout' and 'rtp_timeout_hold' options
are set and RTP is not received that the channel is hung up.
ASTERISK-25259 #close
Change-Id: I85a8cf7c9858ab41d6708bc4b6476bef815f45b5
---
M tests/channels/pjsip/rtp/tests.yaml
A tests/channels/pjsip/rtp/timeout/configs/ast1/extensions.conf
A tests/channels/pjsip/rtp/timeout/configs/ast1/pjsip.conf
A tests/channels/pjsip/rtp/timeout/sipp/uac.xml
A tests/channels/pjsip/rtp/timeout/test-config.yaml
A tests/channels/pjsip/rtp/timeout_hold/configs/ast1/extensions.conf
A tests/channels/pjsip/rtp/timeout_hold/configs/ast1/pjsip.conf
A tests/channels/pjsip/rtp/timeout_hold/sipp/uac.xml
A tests/channels/pjsip/rtp/timeout_hold/test-config.yaml
9 files changed, 405 insertions(+), 0 deletions(-)
Approvals:
Kevin Harwell: Looks good to me, but someone else must approve
Matt Jordan: Looks good to me, approved; Verified
diff --git a/tests/channels/pjsip/rtp/tests.yaml b/tests/channels/pjsip/rtp/tests.yaml
index 6757b6a..5d77214 100644
--- a/tests/channels/pjsip/rtp/tests.yaml
+++ b/tests/channels/pjsip/rtp/tests.yaml
@@ -1,3 +1,5 @@
# Enter tests here in the order they should be considered for execution:
tests:
- dir: 'rtp_keepalive'
+ - test: 'timeout'
+ - test: 'timeout_hold'
diff --git a/tests/channels/pjsip/rtp/timeout/configs/ast1/extensions.conf b/tests/channels/pjsip/rtp/timeout/configs/ast1/extensions.conf
new file mode 100644
index 0000000..cdc2821
--- /dev/null
+++ b/tests/channels/pjsip/rtp/timeout/configs/ast1/extensions.conf
@@ -0,0 +1,4 @@
+[default]
+exten => test,1,Answer()
+same => n,Playback(demo-congrats)
+same => n,Hangup()
diff --git a/tests/channels/pjsip/rtp/timeout/configs/ast1/pjsip.conf b/tests/channels/pjsip/rtp/timeout/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..f54a8e2
--- /dev/null
+++ b/tests/channels/pjsip/rtp/timeout/configs/ast1/pjsip.conf
@@ -0,0 +1,9 @@
+[main-transport]
+type=transport
+protocol=udp
+bind=127.0.0.1
+
+[sipp]
+type=endpoint
+allow=ulaw
+rtp_timeout=5
diff --git a/tests/channels/pjsip/rtp/timeout/sipp/uac.xml b/tests/channels/pjsip/rtp/timeout/sipp/uac.xml
new file mode 100644
index 0000000..4436389
--- /dev/null
+++ b/tests/channels/pjsip/rtp/timeout/sipp/uac.xml
@@ -0,0 +1,119 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- This program is free software; you can redistribute it and/or -->
+<!-- modify it under the terms of the GNU General Public License as -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version. -->
+<!-- -->
+<!-- This program is distributed in the hope that it will be useful, -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
+<!-- GNU General Public License for more details. -->
+<!-- -->
+<!-- You should have received a copy of the GNU General Public License -->
+<!-- along with this program; if not, write to the -->
+<!-- Free Software Foundation, Inc., -->
+<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
+<!-- -->
+<!-- Sipp default 'uac' scenario. -->
+<!-- -->
+
+<scenario name="Basic Sipstone UAC">
+ <!-- In client mode (sipp placing calls), the Call-ID MUST be -->
+ <!-- generated by sipp. To do so, use [call_id] keyword. -->
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:test@[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:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ 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>
+
+ <!-- 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. -->
+ <recv response="200" rtd="true">
+ </recv>
+
+ <!-- Packet lost can be simulated in any send/recv message by -->
+ <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
+ <send>
+ <![CDATA[
+
+ ACK sip:test@[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:test@[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>
+
+ <recv request="BYE">
+ </recv>
+
+ <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>
+
+ <!-- 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. -->
+ <pause 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/rtp/timeout/test-config.yaml b/tests/channels/pjsip/rtp/timeout/test-config.yaml
new file mode 100644
index 0000000..639f14a
--- /dev/null
+++ b/tests/channels/pjsip/rtp/timeout/test-config.yaml
@@ -0,0 +1,44 @@
+testinfo:
+ summary: 'Ensure channel is hung up when RTP timeout is reached.'
+ description: |
+ 'A SIPp scenario places a call into Asterisk. The call is immediately answered and
+ media is sent to the address provided in the SDP. SIPp is not configured to echo
+ or send RTP. The endpoint the SIPp scenario uses is configured to have an RTP timeout
+ of 5 seconds. After 5 seconds the channel should be hung up as a result of the
+ RTP timeout.'
+
+test-modules:
+ test-object:
+ config-section: sipp-config
+ typename: 'sipp.SIPpTestCase'
+ modules:
+ -
+ config-section: 'ami-config'
+ typename: 'ami.AMIEventModule'
+
+sipp-config:
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'uac.xml'}}
+
+ami-config:
+ -
+ type: 'headermatch'
+ id: '0'
+ conditions:
+ match:
+ Event: 'SoftHangupRequest'
+ Channel: 'PJSIP/sipp-*'
+ count: '2'
+
+properties:
+ minversion: '13.5.0'
+ dependencies:
+ - python: 'twisted'
+ - python: 'starpy'
+ - app: 'sipp'
+ - asterisk: 'res_pjsip'
+ - asterisk: 'res_pjsip_session'
+ tags:
+ - pjsip
diff --git a/tests/channels/pjsip/rtp/timeout_hold/configs/ast1/extensions.conf b/tests/channels/pjsip/rtp/timeout_hold/configs/ast1/extensions.conf
new file mode 100644
index 0000000..cdc2821
--- /dev/null
+++ b/tests/channels/pjsip/rtp/timeout_hold/configs/ast1/extensions.conf
@@ -0,0 +1,4 @@
+[default]
+exten => test,1,Answer()
+same => n,Playback(demo-congrats)
+same => n,Hangup()
diff --git a/tests/channels/pjsip/rtp/timeout_hold/configs/ast1/pjsip.conf b/tests/channels/pjsip/rtp/timeout_hold/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..1de9f9c
--- /dev/null
+++ b/tests/channels/pjsip/rtp/timeout_hold/configs/ast1/pjsip.conf
@@ -0,0 +1,9 @@
+[main-transport]
+type=transport
+protocol=udp
+bind=127.0.0.1
+
+[sipp]
+type=endpoint
+allow=ulaw
+rtp_timeout_hold=5
diff --git a/tests/channels/pjsip/rtp/timeout_hold/sipp/uac.xml b/tests/channels/pjsip/rtp/timeout_hold/sipp/uac.xml
new file mode 100644
index 0000000..94c47d6
--- /dev/null
+++ b/tests/channels/pjsip/rtp/timeout_hold/sipp/uac.xml
@@ -0,0 +1,170 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- This program is free software; you can redistribute it and/or -->
+<!-- modify it under the terms of the GNU General Public License as -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version. -->
+<!-- -->
+<!-- This program is distributed in the hope that it will be useful, -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
+<!-- GNU General Public License for more details. -->
+<!-- -->
+<!-- You should have received a copy of the GNU General Public License -->
+<!-- along with this program; if not, write to the -->
+<!-- Free Software Foundation, Inc., -->
+<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
+<!-- -->
+<!-- Sipp default 'uac' scenario. -->
+<!-- -->
+
+<scenario name="Basic Sipstone UAC">
+ <!-- In client mode (sipp placing calls), the Call-ID MUST be -->
+ <!-- generated by sipp. To do so, use [call_id] keyword. -->
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:test@[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:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ 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>
+
+ <!-- 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. -->
+ <recv response="200" rtd="true">
+ </recv>
+
+ <!-- Packet lost can be simulated in any send/recv message by -->
+ <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
+ <send>
+ <![CDATA[
+
+ ACK sip:test@[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:test@[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>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:test@[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:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 INVITE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ 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
+ a=sendonly
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:test@[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:test@[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">
+ </recv>
+
+ <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>
+
+ <!-- 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. -->
+ <pause 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/rtp/timeout_hold/test-config.yaml b/tests/channels/pjsip/rtp/timeout_hold/test-config.yaml
new file mode 100644
index 0000000..55a7b07
--- /dev/null
+++ b/tests/channels/pjsip/rtp/timeout_hold/test-config.yaml
@@ -0,0 +1,44 @@
+testinfo:
+ summary: 'Ensure channel is hung up when RTP hold timeout is reached.'
+ description: |
+ 'A SIPp scenario places a call into Asterisk. The call is immediately answered and
+ media is sent to the address provided in the SDP. SIPp then places the call on hold.
+ SIPp is not configured to echo or send RTP. The endpoint the SIPp scenario uses is
+ configured to have an RTP hold timeout of 5 seconds. After 5 seconds the channel
+ should be hung up as a result of the RTP hold timeout.'
+
+test-modules:
+ test-object:
+ config-section: sipp-config
+ typename: 'sipp.SIPpTestCase'
+ modules:
+ -
+ config-section: 'ami-config'
+ typename: 'ami.AMIEventModule'
+
+sipp-config:
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'uac.xml'}}
+
+ami-config:
+ -
+ type: 'headermatch'
+ id: '0'
+ conditions:
+ match:
+ Event: 'SoftHangupRequest'
+ Channel: 'PJSIP/sipp-*'
+ count: '2'
+
+properties:
+ minversion: '13.5.0'
+ dependencies:
+ - python: 'twisted'
+ - python: 'starpy'
+ - app: 'sipp'
+ - asterisk: 'res_pjsip'
+ - asterisk: 'res_pjsip_session'
+ tags:
+ - pjsip
--
To view, visit https://gerrit.asterisk.org/939
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Gerrit-MessageType: merged
Gerrit-Change-Id: I85a8cf7c9858ab41d6708bc4b6476bef815f45b5
Gerrit-PatchSet: 2
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>
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