[Asterisk-code-review] res/pjsip Add test for 'auto info' (testsuite[master])
Torrey Searle
asteriskteam at digium.com
Mon Jun 19 08:05:16 CDT 2017
Torrey Searle has uploaded this change for review. ( https://gerrit.asterisk.org/5878
Change subject: res/pjsip Add test for 'auto_info'
......................................................................
res/pjsip Add test for 'auto_info'
test to validate new dtmf mode 'auto_info'
ASTERISK-27066 #close
Change-Id: I674d3a6dc678f275bb39738505ee032dd86dbb37
---
A tests/channels/pjsip/dtmf_info_fallback/configs/ast1/extensions.conf
A tests/channels/pjsip/dtmf_info_fallback/configs/ast1/pjsip.conf
A tests/channels/pjsip/dtmf_info_fallback/run-test
A tests/channels/pjsip/dtmf_info_fallback/sipp/A_PARTY.xml
A tests/channels/pjsip/dtmf_info_fallback/sipp/B_PARTY.xml
A tests/channels/pjsip/dtmf_info_fallback/sipp/dtmf_2833_4.pcap
A tests/channels/pjsip/dtmf_info_fallback/test-config.yaml
M tests/channels/pjsip/tests.yaml
8 files changed, 407 insertions(+), 0 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/78/5878/1
diff --git a/tests/channels/pjsip/dtmf_info_fallback/configs/ast1/extensions.conf b/tests/channels/pjsip/dtmf_info_fallback/configs/ast1/extensions.conf
new file mode 100644
index 0000000..35c0a08
--- /dev/null
+++ b/tests/channels/pjsip/dtmf_info_fallback/configs/ast1/extensions.conf
@@ -0,0 +1,12 @@
+[general]
+static=yes
+writeprotect=yes
+autofallthrough=yes
+clearglobalvars=no
+priorityjumping=yes
+
+[globals]
+
+[default]
+exten => s,n,Dial(pjsip/sbc,180)
+
diff --git a/tests/channels/pjsip/dtmf_info_fallback/configs/ast1/pjsip.conf b/tests/channels/pjsip/dtmf_info_fallback/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..025b2a9
--- /dev/null
+++ b/tests/channels/pjsip/dtmf_info_fallback/configs/ast1/pjsip.conf
@@ -0,0 +1,80 @@
+;--
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements start
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[general]
+sipdebug = yes
+
+[PEER_A]
+port = 6060
+
+[sbc]
+port = 5700
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements end
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+--;
+
+
+[global]
+type = global
+debug = yes
+
+[transport-udp6]
+type = transport
+protocol = udp
+bind = [::]:5060
+
+[transport-udp]
+type = transport
+protocol = udp
+bind = 0.0.0.0:5060
+
+[PEER_A]
+type = aor
+contact = sip:127.0.0.1:6060
+
+[PEER_A]
+type = identify
+endpoint = PEER_A
+match = 127.0.0.1
+
+[PEER_A]
+type = endpoint
+context = default
+dtmf_mode = auto_info
+disallow = all
+allow = alaw
+allow = ulaw
+allow = g729
+allow = h263p
+allow = h264
+direct_media = no
+send_rpid = yes
+sdp_session = session
+t38_udptl = yes
+t38_udptl_ec = redundancy
+
+[sbc]
+type = aor
+contact = sip:127.0.0.1:5700
+
+[sbc]
+type = endpoint
+context = callcontrol
+dtmf_mode = auto_info
+disallow = all
+allow = alaw
+allow = ulaw
+allow = g729
+allow = h263p
+allow = h264
+direct_media = no
+send_rpid = yes
+sdp_session = session
+aors = sbc
+t38_udptl = yes
+t38_udptl_ec = redundancy
+
diff --git a/tests/channels/pjsip/dtmf_info_fallback/run-test b/tests/channels/pjsip/dtmf_info_fallback/run-test
new file mode 100755
index 0000000..241bd58
--- /dev/null
+++ b/tests/channels/pjsip/dtmf_info_fallback/run-test
@@ -0,0 +1,70 @@
+#!/usr/bin/env python
+'''
+Copyright (C) 2010, Digium, Inc.
+Russell Bryant <russell at digium.com>
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+'''
+
+import sys
+import os
+import signal
+import subprocess
+import time
+
+sys.path.append("lib/python")
+sys.path.append("utils")
+
+from twisted.internet import reactor
+from asterisk.sipp import SIPpTest
+
+WORKING_DIR = os.path.abspath(os.path.dirname(__file__))
+TEST_DIR = os.path.dirname(os.path.realpath(__file__))
+
+sippA_logfile = WORKING_DIR + "/A_PARTY.log"
+sippA_errfile = WORKING_DIR + "/A_PARTY_ERR.log"
+sippB_logfile = WORKING_DIR + "/B_PARTY.log"
+sippB_errfile = WORKING_DIR + "/B_PARTY_ERR.log"
+SIPP_SCENARIOS = [
+ {
+ 'scenario' : 'B_PARTY.xml',
+ '-i' : '127.0.0.1',
+ '-p' : '5700',
+ '-mp' : '6300',
+ '-message_file' : sippB_logfile,
+ '-error_file' : sippB_errfile,
+ '-trace_msg' : '-trace_err',
+ },
+ {
+ 'scenario' : 'A_PARTY.xml',
+ '-i' : '127.0.0.1',
+ '-p' : '6060',
+ '-s' : '3228080970',
+ '-d' : '35000',
+ '-message_file' : sippA_logfile,
+ '-error_file' : sippA_errfile,
+ '-trace_msg' : '-trace_err',
+ }
+]
+
+
+def main():
+ test = SIPpTest(WORKING_DIR, TEST_DIR, SIPP_SCENARIOS)
+ test.reactor_timeout = 70;
+
+ time.sleep(10) #Wait 5 seconds to ensure that all the sockets are open before running the test
+
+ reactor.run()
+
+ if not test.passed:
+ return 1
+
+ return 0
+
+
+if __name__ == "__main__":
+ sys.exit(main())
+
+
+# vim:sw=4:ts=4:expandtab:textwidth=79
diff --git a/tests/channels/pjsip/dtmf_info_fallback/sipp/A_PARTY.xml b/tests/channels/pjsip/dtmf_info_fallback/sipp/A_PARTY.xml
new file mode 100644
index 0000000..ada8590
--- /dev/null
+++ b/tests/channels/pjsip/dtmf_info_fallback/sipp/A_PARTY.xml
@@ -0,0 +1,124 @@
+<?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 'uac' scenario with pcap (rtp) play -->
+<!-- -->
+<scenario name="DTMF_INFO_FALLBACK">
+ <!-- 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:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test at voxbone.com>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com:[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[local_ip_type] [local_ip]
+ t=0 0
+ m=audio 9000 RTP/AVP 0 101
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ a=fmtp:101 0-15
+
+ ]]>
+ </send>
+
+ <recv response="100" optional="true">
+ </recv>
+
+ <recv response="180" 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" crlf="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:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com>[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>
+
+ <!-- Play a PCAP which sends the RTPEVENT packet containing DTMF 4 -->
+ <nop>
+ <action>
+ <exec play_pcap_audio="./tests/channels/pjsip/dtmf_info_fallback/sipp/dtmf_2833_4.pcap"/>
+ </action>
+ </nop>
+
+ <pause milliseconds="7000"/>
+
+ <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true">
+ </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. -->
+ <pause milliseconds="3000"/>
+
+
+ <!-- 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/dtmf_info_fallback/sipp/B_PARTY.xml b/tests/channels/pjsip/dtmf_info_fallback/sipp/B_PARTY.xml
new file mode 100644
index 0000000..a8a3d92
--- /dev/null
+++ b/tests/channels/pjsip/dtmf_info_fallback/sipp/B_PARTY.xml
@@ -0,0 +1,109 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="DTMF Handling">
+
+<recv request="INVITE" crlf="true">
+</recv>
+
+
+
+<send>
+<![CDATA[
+
+SIP/2.0 100 Trying
+[last_Via:]
+[last_Call-ID:]
+[last_From:]
+[last_To:]
+[last_CSeq:]
+Content-Length: 0
+
+]]>
+</send>
+
+
+<send retrans="500">
+<![CDATA[
+
+SIP/2.0 200 OK
+[last_Via:]
+[last_Call-ID:]
+[last_From:]
+[last_To:];tag=[call_number]
+[last_CSeq:]
+Contact: <sip:736596371553211257@[local_ip]:[local_port];user=phone>
+Content-Type: application/sdp
+Content-Length: [len]
+
+v=0
+o=HuaweiSoftX3000 6644052 6644052 IN IP[local_ip_type] [local_ip]
+s=Sip Call
+c=IN IP[local_ip_type] [local_ip]
+t=0 0
+m=audio 8000 RTP/AVP 0
+a=rtpmap:0 PCMU/8000
+
+]]>
+</send>
+
+
+<recv request="ACK"
+ rtd="true"
+ crlf="true">
+
+</recv>
+
+<!-- Receive the DIGIT 4-->
+<recv request="INFO">
+ <action>
+ <ereg regexp="(Signal=4)" search_in="body" check_it="true" assign_to = "1" />
+ <log message="---DTMF--- [$1]"/>
+ </action>
+</recv>
+
+<send>
+<![CDATA[
+
+SIP/2.0 200 OK
+[last_Via:]
+[last_From:]
+[last_To:]
+[last_Call-ID:]
+[last_CSeq:]
+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:]
+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="3000"/>
+
+
+<!-- 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/dtmf_info_fallback/sipp/dtmf_2833_4.pcap b/tests/channels/pjsip/dtmf_info_fallback/sipp/dtmf_2833_4.pcap
new file mode 100644
index 0000000..b3bd1ef
--- /dev/null
+++ b/tests/channels/pjsip/dtmf_info_fallback/sipp/dtmf_2833_4.pcap
Binary files differ
diff --git a/tests/channels/pjsip/dtmf_info_fallback/test-config.yaml b/tests/channels/pjsip/dtmf_info_fallback/test-config.yaml
new file mode 100644
index 0000000..e0c5c52
--- /dev/null
+++ b/tests/channels/pjsip/dtmf_info_fallback/test-config.yaml
@@ -0,0 +1,11 @@
+testinfo:
+ summary: 'This test case verifies the DTMF INFO FALLBACK i.e fallback tO INFO instead of INBAND'
+ description: |
+ 'This test case verifies the DTMF INFO FALLBACK i.e fallback tO INFO instead of INBAND'
+
+properties:
+ minversion: '13'
+ dependencies:
+ - app : 'sipp'
+ tags:
+ - SIP
diff --git a/tests/channels/pjsip/tests.yaml b/tests/channels/pjsip/tests.yaml
index 59f5f1c..54ff285 100644
--- a/tests/channels/pjsip/tests.yaml
+++ b/tests/channels/pjsip/tests.yaml
@@ -53,3 +53,4 @@
- test: 'user_eq_phone'
- test: 'cseq_method'
- test: 'multipart_empty_part'
+ - test: 'dtmf_info_fallback'
--
To view, visit https://gerrit.asterisk.org/5878
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-MessageType: newchange
Gerrit-Change-Id: I674d3a6dc678f275bb39738505ee032dd86dbb37
Gerrit-Change-Number: 5878
Gerrit-PatchSet: 1
Gerrit-Owner: Torrey Searle <tsearle at gmail.com>
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