[svn-commits] mjordan: testsuite/asterisk/trunk r5102 - in /asterisk/trunk/tests/channels/p...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Sun Jun 8 13:09:20 CDT 2014
Author: mjordan
Date: Sun Jun 8 13:09:09 2014
New Revision: 5102
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=5102
Log:
channels/pjsip/transfers/blind_transfer: Update direct media tests
This patch updates the direct media tests for the expected re-INVITEs that will
occur during proper handling of a SIP blind transfer that involve two direct
media bridges. This includes:
- For the party in the bridge with the transfer destination, a re-INVITE that
redirects media back to Asterisk will occur nearly at the same time as a BYE
request (the order from Asterisk will always be re-INVITE, then BYE). This
handles this situation by caching the necessary values from the re-INVITE,
handling the BYE, then sending the response to the re-INVITE.
- A previously commented out direct media re-INVITE was restored. It should
never have been commented out.
Review: https://reviewboard.asterisk.org/r/3586/
Modified:
asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/callee_direct_media/sipp/alice.xml
asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/callee_direct_media/sipp/bob.xml
asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/caller_direct_media/sipp/alice.xml
asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/caller_direct_media/sipp/bob.xml
Modified: asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/callee_direct_media/sipp/alice.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/callee_direct_media/sipp/alice.xml?view=diff&rev=5102&r1=5101&r2=5102
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/callee_direct_media/sipp/alice.xml (original)
+++ asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/callee_direct_media/sipp/alice.xml Sun Jun 8 13:09:09 2014
@@ -2,7 +2,7 @@
<scenario name="Send Call">
- <send retrans="500">
+ <send retrans="500" start_txn="initial_invite">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
@@ -27,13 +27,13 @@
]]>
</send>
- <recv response="100" optional="true" />
- <recv response="180" optional="true" />
- <recv response="183" optional="true" />
-
- <recv response="200" rtd="true" />
-
- <send>
+ <recv response="100" optional="true" response_txn="initial_invite"/>
+ <recv response="180" optional="true" response_txn="initial_invite"/>
+ <recv response="183" optional="true" response_txn="initial_invite"/>
+
+ <recv response="200" rtd="true" response_txn="initial_invite"/>
+
+ <send ack_txn="initial_invite">
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
@@ -155,19 +155,68 @@
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:]
+ <!--
+ A re-INVITE will come from Asterisk to redirect media back to it
+ as Alice leaves the bridge. At the same time, Alice will get hung up.
+ Cache the needed values from the INVITE so we can handle both requets.
+ -->
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp="(.*)"
+ header="Via:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="reinvite_via"/>
+ <ereg regexp="(.*)"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="reinvite_from"/>
+ <ereg regexp="(.*)"
+ header="To:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="reinvite_to"/>
+ <ereg regexp="(.*)"
+ header="Call-ID:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="reinvite_call_id"/>
+ <ereg regexp="(.*)"
+ header="CSeq:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="reinvite_cseq"/>
+ </action>
+ </recv>
+
+ <recv request="BYE" />
+
+ <send>
+ <![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: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ Via: [$reinvite_via]
+ From: [$reinvite_from]
+ To: [$reinvite_to]
+ Call-ID: [$reinvite_call_id]
+ CSeq: [$reinvite_cseq]
Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
@@ -188,23 +237,6 @@
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"/>
Modified: asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/callee_direct_media/sipp/bob.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/callee_direct_media/sipp/bob.xml?view=diff&rev=5102&r1=5101&r2=5102
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/callee_direct_media/sipp/bob.xml (original)
+++ asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/callee_direct_media/sipp/bob.xml Sun Jun 8 13:09:09 2014
@@ -146,7 +146,7 @@
Wait for re-invite for tearing down direct media between alice and bob
due to transfering alice to charlie.
-->
- <!-- <recv request="INVITE" crlf="true"/>
+ <recv request="INVITE" crlf="true"/>
<send retrans="500" next="11">
<![CDATA[
@@ -179,7 +179,7 @@
crlf="true"
next="12" >
</recv>
--->
+
<label id="12" />
<!-- We should receive a NOTIFY from Asterisk with a '200 OK' sipfrag -->
Modified: asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/caller_direct_media/sipp/alice.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/caller_direct_media/sipp/alice.xml?view=diff&rev=5102&r1=5101&r2=5102
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/caller_direct_media/sipp/alice.xml (original)
+++ asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/caller_direct_media/sipp/alice.xml Sun Jun 8 13:09:09 2014
@@ -2,7 +2,7 @@
<scenario name="Send Call">
- <send retrans="500">
+ <send retrans="500" start_txn="initial_invite">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
@@ -27,11 +27,11 @@
]]>
</send>
- <recv response="100" optional="true" />
- <recv response="180" optional="true" />
- <recv response="183" optional="true" />
-
- <recv response="200" rtd="true">
+ <recv response="100" optional="true" response_txn="initial_invite"/>
+ <recv response="180" optional="true" response_txn="initial_invite"/>
+ <recv response="183" optional="true" response_txn="initial_invite"/>
+
+ <recv response="200" rtd="true" response_txn="initial_invite">
<!-- Save the To tag. We'll need it when we send REFER -->
<action>
<ereg regexp="(;tag=.*)"
@@ -42,7 +42,7 @@
</action>
</recv>
- <send>
+ <send ack_txn="initial_invite">
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
@@ -146,7 +146,7 @@
Wait for re-invite for tearing down direct media between alice and bob
due to transfering bob to charlie.
-->
-<!-- <recv request="INVITE" crlf="true"/>
+ <recv request="INVITE" crlf="true"/>
<send retrans="500">
<![CDATA[
@@ -176,7 +176,7 @@
rtd="true"
crlf="true">
</recv>
--->
+
<!-- We should receive a NOTIFY from Asterisk with a '200 OK' sipfrag -->
<recv request="NOTIFY" crlf="true">
<!-- Check if 'SIP/2.0 200 OK' is in the body -->
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=5102&r1=5101&r2=5102
==============================================================================
--- 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 Jun 8 13:09:09 2014
@@ -155,19 +155,68 @@
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:]
+ <!--
+ A re-INVITE will come from Asterisk to redirect media back to it
+ as Bob leaves the bridge. At the same time, Bob will get hung up.
+ Cache the needed values from the INVITE so we can handle both requets.
+ -->
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp="(.*)"
+ header="Via:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="reinvite_via"/>
+ <ereg regexp="(.*)"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="reinvite_from"/>
+ <ereg regexp="(.*)"
+ header="To:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="reinvite_to"/>
+ <ereg regexp="(.*)"
+ header="Call-ID:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="reinvite_call_id"/>
+ <ereg regexp="(.*)"
+ header="CSeq:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="reinvite_cseq"/>
+ </action>
+ </recv>
+
+ <recv request="BYE" />
+
+ <send>
+ <![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: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ Via: [$reinvite_via]
+ From: [$reinvite_from]
+ To: [$reinvite_to]
+ Call-ID: [$reinvite_call_id]
+ CSeq: [$reinvite_cseq]
Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
@@ -188,22 +237,6 @@
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"/>
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