[Asterisk-code-review] AST-2021-007 - pjsip: Test for receiving re-INVITE after sending BYE. (testsuite[18])
Friendly Automation
asteriskteam at digium.com
Fri Jul 23 10:57:11 CDT 2021
Friendly Automation has uploaded this change for review. ( https://gerrit.asterisk.org/c/testsuite/+/16198 )
Change subject: AST-2021-007 - pjsip: Test for receiving re-INVITE after sending BYE.
......................................................................
AST-2021-007 - pjsip: Test for receiving re-INVITE after sending BYE.
This adds a test that confirms upon receiving a re-INVITE after
we have sent a BYE that we do not crash.
ASTERISK-29381
Change-Id: I39f71f076a1d15a64f5f1efa0b2d5692c456b331
---
A tests/channels/pjsip/reinvite_after_bye/configs/ast1/extensions.conf
A tests/channels/pjsip/reinvite_after_bye/configs/ast1/pjsip.conf
A tests/channels/pjsip/reinvite_after_bye/sipp/reinvite-after-bye.xml
A tests/channels/pjsip/reinvite_after_bye/test-config.yaml
M tests/channels/pjsip/tests.yaml
5 files changed, 186 insertions(+), 0 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/98/16198/1
diff --git a/tests/channels/pjsip/reinvite_after_bye/configs/ast1/extensions.conf b/tests/channels/pjsip/reinvite_after_bye/configs/ast1/extensions.conf
new file mode 100644
index 0000000..7a5b80a
--- /dev/null
+++ b/tests/channels/pjsip/reinvite_after_bye/configs/ast1/extensions.conf
@@ -0,0 +1,4 @@
+[default]
+exten => test,1,Answer()
+same => n,Wait(0.1)
+same => n,Hangup
diff --git a/tests/channels/pjsip/reinvite_after_bye/configs/ast1/pjsip.conf b/tests/channels/pjsip/reinvite_after_bye/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..68bcd4b
--- /dev/null
+++ b/tests/channels/pjsip/reinvite_after_bye/configs/ast1/pjsip.conf
@@ -0,0 +1,14 @@
+[system]
+type=system
+timer_t1=100
+timer_b=6400
+
+[main-transport]
+type=transport
+protocol=udp
+bind=127.0.0.1
+
+[sipp]
+type=endpoint
+send_pai=yes
+allow=ulaw
diff --git a/tests/channels/pjsip/reinvite_after_bye/sipp/reinvite-after-bye.xml b/tests/channels/pjsip/reinvite_after_bye/sipp/reinvite-after-bye.xml
new file mode 100644
index 0000000..7426f37
--- /dev/null
+++ b/tests/channels/pjsip/reinvite_after_bye/sipp/reinvite-after-bye.xml
@@ -0,0 +1,137 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="491 Test">
+ <!-- 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:[service]@[remote_ip]>
+ 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:8 PCMA/8000
+ a=rtpmap:0 PCMU/8000
+ a=ptime:20
+
+ ]]>
+ </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>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:[service]@[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:[service]@[remote_ip]>[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" crlf="true" />
+
+ <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:[service]@[remote_ip]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 INVITE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:[service]@[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:[service]@[remote_ip]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 ACK
+ 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:8 PCMA/8000
+ a=rtpmap:0 PCMU/8000
+ a=ptime:20
+
+ ]]>
+ </send>
+
+ <recv request="BYE" crlf="true" />
+
+ <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:alice@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, MESSAGE, BYE
+ Content-Type: application/sdp
+ Content-Length: 0
+ ]]>
+ </send>
+
+</scenario>
+
diff --git a/tests/channels/pjsip/reinvite_after_bye/test-config.yaml b/tests/channels/pjsip/reinvite_after_bye/test-config.yaml
new file mode 100644
index 0000000..15cad50
--- /dev/null
+++ b/tests/channels/pjsip/reinvite_after_bye/test-config.yaml
@@ -0,0 +1,30 @@
+testinfo:
+ summary: 'Ensure graceful operation when receiving a re-INVITE after sending a BYE.'
+ description: |
+ 'A SIPp scenario places a call into Asterisk. Once the call has been answered, we
+ immediately hang it up causing a BYE to be sent to the caller. Upon the caller
+ receiving the BYE it sends a re-INVITE without SDP instead of responding to the BYE.
+ The re-INVITE is handled like normal and once the caller receives a retransmission
+ of the BYE it then handles the BYE like normal.'
+
+test-modules:
+ test-object:
+ config-section: sipp-config
+ typename: 'sipp.SIPpTestCase'
+
+sipp-config:
+ memcheck-delay-stop: 7
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'reinvite-after-bye.xml'}}
+
+properties:
+ dependencies:
+ - python: 'twisted'
+ - python: 'starpy'
+ - app: 'sipp'
+ - asterisk: 'res_pjsip'
+ - asterisk: 'res_pjsip_session'
+ tags:
+ - pjsip
diff --git a/tests/channels/pjsip/tests.yaml b/tests/channels/pjsip/tests.yaml
index 280a4a0..1b4fad8 100644
--- a/tests/channels/pjsip/tests.yaml
+++ b/tests/channels/pjsip/tests.yaml
@@ -63,3 +63,4 @@
- test: 'moh_passthru_inactive'
- test: 'non_negotiated_frame_SSRC_change'
- test: 'content_disposition'
+ - test: 'reinvite_after_bye'
--
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Gerrit-Project: testsuite
Gerrit-Branch: 18
Gerrit-Change-Id: I39f71f076a1d15a64f5f1efa0b2d5692c456b331
Gerrit-Change-Number: 16198
Gerrit-PatchSet: 1
Gerrit-Owner: Friendly Automation
Gerrit-CC: Joshua Colp <jcolp at sangoma.com>
Gerrit-MessageType: newchange
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