[asterisk-commits] mjordan: testsuite/asterisk/trunk r5049 - in /asterisk/trunk/tests/channels/p...

[SVN commits to the Asterisk project]

Author: mjordan
Date: Sun May 18 15:30:58 2014
New Revision: 5049

URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=5049
Log:
pjsip/transfers/blind_transfer/caller_direct_media: Fix race conditions

This patch fixes a few race conditions in the test where NOTIFY requests
could be sent interleaved with direct media re-INVITE requests. It also
makes it expect some direct media requests when channels are ejected from
a bridge.

Modified:
    asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/caller_direct_media/configs/ast1/extensions.conf
    asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/caller_direct_media/sipp/bob.xml

Modified: asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/caller_direct_media/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/caller_direct_media/configs/ast1/extensions.conf?view=diff&rev=5049&r1=5048&r2=5049
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/caller_direct_media/configs/ast1/extensions.conf (original)
+++ asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/caller_direct_media/configs/ast1/extensions.conf Sun May 18 15:30:58 2014
@@ -13,13 +13,25 @@
 	same => n,Hangup()
 
 exten => charlie,1,NoOp()
+	; The Wait here is annoying, but necessary. We need to
+	; allow some time to pass before dialing Charlie, otherwise
+	; Charlie's answering will cause a NOTIFY request to be sent
+	; prior to the direct media INVITE requests being fully
+	; processed
+	same => n,Wait(5)
 	same => n,Dial(PJSIP/charlie)
 	same => n,Hangup()
 
 [other]
 ; Second test iteration should execute
 exten => charlie,1,NoOp()
-        same => n,Dial(PJSIP/charlie)
+	; The Wait here is annoying, but necessary. We need to
+	; allow some time to pass before dialing Charlie, otherwise
+	; Charlie's answering will cause a NOTIFY request to be sent
+	; prior to the direct media INVITE requests being fully
+	; processed
+	same => n,Wait(5)
+    same => n,Dial(PJSIP/charlie)
 	same => n,Hangup()
 
 

Modified: asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/caller_direct_media/sipp/bob.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/caller_direct_media/sipp/bob.xml?view=diff&rev=5049&r1=5048&r2=5049
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/caller_direct_media/sipp/bob.xml (original)
+++ asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/caller_direct_media/sipp/bob.xml Sun May 18 15:30:58 2014
@@ -155,6 +155,34 @@
         crlf="true">
   </recv>
 
+  <!-- Wait for re-invite to setup RTP directly between bob and Asterisk
+       after charlie hangs up -->
+  <recv request="INVITE" crlf="true"/>
+
+  <send retrans="500">
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:]
+      [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 [custom_media_port] RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
   <recv request="BYE" />
 
   <send>