[Asterisk-code-review] tests/pjsip/hold: Allow optional reinvite. (testsuite[master])

[Friendly Automation]

Friendly Automation has submitted this change and it was merged. ( https://gerrit.asterisk.org/10900 )

Change subject: tests/pjsip/hold: Allow optional reinvite.
......................................................................

tests/pjsip/hold: Allow optional reinvite.

Due to a race condition between SDP negotiation and bridging
it is possible for a reinvite to be sent to the calling party
in this test. The test did not allow this, causing a failure
to occur.

This change adds support for this optional reinvite.

This is part of my continuing saga to better understand the
test failures we're seeing.

Change-Id: Ia267cc138966dbd4449f2cab0fa56c01819a8c90
---
M tests/channels/pjsip/hold/sipp/phone_A.xml
1 file changed, 40 insertions(+), 1 deletion(-)

Approvals:
  Kevin Harwell: Looks good to me, but someone else must approve
  George Joseph: Looks good to me, approved
  Friendly Automation: Approved for Submit



diff --git a/tests/channels/pjsip/hold/sipp/phone_A.xml b/tests/channels/pjsip/hold/sipp/phone_A.xml
index 8c2ae2b..bb404a3 100644
--- a/tests/channels/pjsip/hold/sipp/phone_A.xml
+++ b/tests/channels/pjsip/hold/sipp/phone_A.xml
@@ -58,9 +58,13 @@
 		]]>
 	</send>
 
+	<recv request="INVITE" optional="true" next="respond_reinvite"/>
+
+	<label id="wait_for_bye"/>
+
 	<recv request="BYE"/>
 
-	<send retrans="500">
+	<send retrans="500" next="scenario_end">
 		<![CDATA[
 			SIP/2.0 200 OK
 			[last_Via:]
@@ -89,4 +93,39 @@
 		]]>
 	</send>
 
+	<label id="respond_reinvite"/>
+
+	<send retrans="500" next="scenario_end">
+                <![CDATA[
+                        SIP/2.0 200 OK
+                        [last_Via:]
+                        [last_From:]
+                        [last_To:];tag=[call_number]
+                        [last_Call-ID:]
+                        [last_CSeq:]
+                        Contact: <sip:[field0]@[local_ip]:[local_port];transport=[transport]>
+                        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+                        Supported: 100rel,replaces
+                        User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+                        Accept-Language: en
+                        Content-Type: application/sdp
+                        Content-Length: [len]
+
+                        v=0
+                        o=- 1324901698 1324901698 IN IP4 [local_ip]
+                        s=Polycom IP Phone
+                        c=IN IP4 [local_ip]
+                        t=0 0
+                        a=sendrecv
+                        m=audio 2226 RTP/AVP 0 101
+                        a=sendrecv
+                        a=rtpmap:0 PCMU/8000
+                        a=rtpmap:101 telephone-event/8000
+                ]]>
+	</send>
+
+	<recv request="ACK" next="wait_for_bye"/>
+
+	<label id="scenario_end"/>
+
 </scenario>

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Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-MessageType: merged
Gerrit-Change-Id: Ia267cc138966dbd4449f2cab0fa56c01819a8c90
Gerrit-Change-Number: 10900
Gerrit-PatchSet: 1
Gerrit-Owner: Joshua C. Colp <jcolp at digium.com>
Gerrit-Reviewer: Friendly Automation (1000185)
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua C. Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
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