[Asterisk-code-review] local: Add test for dialing with a removed/declined audio stream. (testsuite[master])
Friendly Automation
asteriskteam at digium.com
Fri Apr 30 09:37:14 CDT 2021
Friendly Automation has submitted this change. ( https://gerrit.asterisk.org/c/testsuite/+/15837 )
Change subject: local: Add test for dialing with a removed/declined audio stream.
......................................................................
local: Add test for dialing with a removed/declined audio stream.
This change adds a test which dials from PJSIP, to a Local channel,
and then to another PJSIP channel. The incoming call leg offers a
normal audio stream as well as a declined audio stream. The test
confirms that the Local channel is successfully dialed and that the
called PJSIP party receives the call.
ASTERISK-29407
Change-Id: If55c6ef39c322cb45f59e93bd4ef545957cf5c26
---
A tests/channels/local/local_removed_audio_stream_request/configs/ast1/extensions.conf
A tests/channels/local/local_removed_audio_stream_request/configs/ast1/pjsip.conf
A tests/channels/local/local_removed_audio_stream_request/sipp/alice.xml
A tests/channels/local/local_removed_audio_stream_request/sipp/bob.xml
A tests/channels/local/local_removed_audio_stream_request/test-config.yaml
M tests/channels/local/tests.yaml
6 files changed, 230 insertions(+), 0 deletions(-)
Approvals:
Kevin Harwell: Looks good to me, approved
George Joseph: Looks good to me, but someone else must approve
Friendly Automation: Approved for Submit
diff --git a/tests/channels/local/local_removed_audio_stream_request/configs/ast1/extensions.conf b/tests/channels/local/local_removed_audio_stream_request/configs/ast1/extensions.conf
new file mode 100644
index 0000000..f5f4823
--- /dev/null
+++ b/tests/channels/local/local_removed_audio_stream_request/configs/ast1/extensions.conf
@@ -0,0 +1,12 @@
+[general]
+
+[globals]
+
+[calling]
+exten => bob,1,Dial(Local/bob at bob)
+
+[bob]
+exten => bob,1,NoOp()
+ same => n,Dial(PJSIP/bob)
+ same => n,Hangup()
+
diff --git a/tests/channels/local/local_removed_audio_stream_request/configs/ast1/pjsip.conf b/tests/channels/local/local_removed_audio_stream_request/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..c82ab09
--- /dev/null
+++ b/tests/channels/local/local_removed_audio_stream_request/configs/ast1/pjsip.conf
@@ -0,0 +1,34 @@
+[system]
+type=system
+timer_t1=100
+timer_b=6400
+
+[global]
+type=global
+debug=yes
+
+[local-transport]
+type=transport
+bind=127.0.0.1
+protocol=udp
+
+[alice]
+type=endpoint
+allow=g722,ulaw,alaw,h264
+context=calling
+direct_media=no
+media_address=127.0.0.1
+aors=alice
+
+[bob]
+type=endpoint
+allow=g722,ulaw,alaw,h264
+context=calling
+direct_media=no
+media_address=127.0.0.1
+aors=bob
+
+[bob]
+type=aor
+max_contacts=1
+contact=sip:bob at 127.0.0.3:5060\;transport=udp
diff --git a/tests/channels/local/local_removed_audio_stream_request/sipp/alice.xml b/tests/channels/local/local_removed_audio_stream_request/sipp/alice.xml
new file mode 100644
index 0000000..e0c4cb0
--- /dev/null
+++ b/tests/channels/local/local_removed_audio_stream_request/sipp/alice.xml
@@ -0,0 +1,73 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="Send Call">
+
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag[call_number]
+ To: <sip:[service]@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: [cseq] INVITE
+ Contact: <sip:alice@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 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
+ m=audio 0 RTP/AVP 0
+
+ ]]>
+ </send>
+
+ <recv response="100" optional="true" />
+ <recv response="180" optional="true" />
+ <recv response="183" optional="true" />
+
+ <recv response="200" rtd="true" />
+
+ <send>
+ <![CDATA[
+
+ ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ Call-ID: [call_id]
+ CSeq: [cseq] ACK
+ Contact: <sip:alice@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ [last_Via:]4
+ From: <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag[call_number]
+ To: <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: [cseq] BYE
+ Contact: sip:alice@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true">
+ </recv>
+
+</scenario>
diff --git a/tests/channels/local/local_removed_audio_stream_request/sipp/bob.xml b/tests/channels/local/local_removed_audio_stream_request/sipp/bob.xml
new file mode 100644
index 0000000..6a4c463
--- /dev/null
+++ b/tests/channels/local/local_removed_audio_stream_request/sipp/bob.xml
@@ -0,0 +1,73 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic UAS responder">
+
+ <recv request="INVITE" crlf="true" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
+ 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 request="ACK"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <recv request="BYE" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
diff --git a/tests/channels/local/local_removed_audio_stream_request/test-config.yaml b/tests/channels/local/local_removed_audio_stream_request/test-config.yaml
new file mode 100644
index 0000000..e99ea06
--- /dev/null
+++ b/tests/channels/local/local_removed_audio_stream_request/test-config.yaml
@@ -0,0 +1,37 @@
+testinfo:
+ summary: 'Test calling from PJSIP through a Local channel to PJSIP with a removed audio stream'
+ description: |
+ 'A SIPp scenario calls into Asterisk and Asterisk then dials another through a Local
+ channel. The initial call leg contains both an accepted audio stream and a
+ removed/declined stream. The test confirms that the Local channel allows
+ dialing with the removed/declined stream and that the called party
+ receives the call.'
+
+properties:
+ dependencies:
+ - python : 'twisted'
+ - python : 'starpy'
+ - asterisk : 'app_dial'
+ - asterisk : 'res_pjsip'
+ - sipp :
+ version : 'v3.4.1'
+ tags:
+ - pjsip
+
+test-modules:
+ add-test-to-search-path: 'True'
+ test-object:
+ config-section: test-case-config
+ typename: 'sipp.SIPpTestCase'
+
+test-case-config:
+ memcheck-delay-stop: 7
+ connect-ami: 'True'
+ fail-on-any: False
+ test-iterations:
+ -
+ scenarios:
+ # Bob receives call from Alice
+ - { 'key-args': {'scenario': 'bob.xml', '-p': '5060', '-i': '127.0.0.3', '-s': 'alice', '-timeout': '20s', '-mi': '127.0.0.3'} }
+ # Alice calls Bob
+ - { 'key-args': {'scenario': 'alice.xml', '-p': '5060', '-i': '127.0.0.2', '-s': 'bob', '-timeout': '20s', '-mi': '127.0.0.2'} }
diff --git a/tests/channels/local/tests.yaml b/tests/channels/local/tests.yaml
index a009b04..2e1f942 100644
--- a/tests/channels/local/tests.yaml
+++ b/tests/channels/local/tests.yaml
@@ -5,3 +5,4 @@
- test: 'local_holding_bridge'
- test: 'local_optimize_away'
- test: 'local_loop'
+ - test: 'local_removed_audio_stream_request'
--
To view, visit https://gerrit.asterisk.org/c/testsuite/+/15837
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Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Change-Id: If55c6ef39c322cb45f59e93bd4ef545957cf5c26
Gerrit-Change-Number: 15837
Gerrit-PatchSet: 3
Gerrit-Owner: Joshua Colp <jcolp at sangoma.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-MessageType: merged
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