[svn-commits] asanders: testsuite/asterisk/trunk r6483 - in /asterisk/trunk/tests/channels/...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Mar 2 22:13:17 CST 2015
Author: asanders
Date: Mon Mar 2 22:13:14 2015
New Revision: 6483
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=6483
Log:
chan_sip: Asterisk fails to re-activate an inactive media session when an
offer does not contain a=sendrecv
This test is to ensure that Asterisk correctly applies the direction of the
media stream when a=<sendonly|recvonly|inactive|sendrecv> is missing from the
offer's SDP. The expected behavior is for Asterisk to apply "sendrecv" as the
direction of the media stream when no direction attribute is present in an
offer's SDP.
According to RFC 4566 (Section 6. SDP Attributes): "If none of the attributes
"sendonly", "recvonly", "inactive", and "sendrecv" is present, "sendrecv"
SHOULD be assumed as the default for sessions that are not of the conference
type "broadcast" or "H332" [...]"
The test scenario:
1. From Phone A, send an offer to Phone B to establish a call
2. From Phone B, send an offer to Phone A to put the call on hold.
3. Observe that the MOH start event occurs.
4. From Phone B, send an offer to Phone A to 'un-hold' the call (ensure that
the direction attribute from the offer's SDP is omitted)
5. Observe that the MOH stop event occurs.
ASTERISK-24824 #close
Reported By: Ashley Sanders
Review: https://reviewboard.asterisk.org/r/4442/
Added:
asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A_no_direction.xml (with props)
asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_no_direction.xml (with props)
Modified:
asterisk/trunk/tests/channels/SIP/sip_hold/run-test
asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml
asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml
asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml
Modified: asterisk/trunk/tests/channels/SIP/sip_hold/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/run-test?view=diff&rev=6483&r1=6482&r2=6483
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/run-test (original)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/run-test Mon Mar 2 22:13:14 2015
@@ -39,15 +39,19 @@
self.sipp_phone_a_scenarios = [{'scenario':'phone_A.xml','-i':'127.0.0.2','-p':'5060','-inf':INJECT_FILE_BYPASS},
{'scenario':'phone_A.xml','-i':'127.0.0.2','-p':'5060','-inf':INJECT_FILE_BYPASS},
{'scenario':'phone_A.xml','-i':'127.0.0.2','-p':'5060','-inf':INJECT_FILE_BYPASS},
+ {'scenario':'phone_A_no_direction.xml','-i':'127.0.0.2','-p':'5060','-inf':INJECT_FILE_BYPASS},
{'scenario':'phone_A.xml','-i':'127.0.0.2','-p':'5060','-inf':INJECT_FILE_BRIDGE},
{'scenario':'phone_A.xml','-i':'127.0.0.2','-p':'5060','-inf':INJECT_FILE_BRIDGE},
- {'scenario':'phone_A.xml','-i':'127.0.0.2','-p':'5060','-inf':INJECT_FILE_BRIDGE},]
+ {'scenario':'phone_A.xml','-i':'127.0.0.2','-p':'5060','-inf':INJECT_FILE_BRIDGE},
+ {'scenario':'phone_A_no_direction.xml','-i':'127.0.0.2','-p':'5060','-inf':INJECT_FILE_BRIDGE}]
self.sipp_phone_b_scenarios = [{'scenario':'phone_B_media_restrict.xml','-i':'127.0.0.3','-p':'5060','-inf':INJECT_FILE_BYPASS},
{'scenario':'phone_B_IP_restrict.xml','-i':'127.0.0.3','-p':'5060','-inf':INJECT_FILE_BYPASS},
{'scenario':'phone_B_IP_media_restrict.xml','-i':'127.0.0.3','-p':'5060','-inf':INJECT_FILE_BYPASS},
+ {'scenario':'phone_B_no_direction.xml','-i':'127.0.0.3','-p':'5060','-inf':INJECT_FILE_BYPASS},
{'scenario':'phone_B_media_restrict.xml','-i':'127.0.0.3','-p':'5060','-inf':INJECT_FILE_BRIDGE},
{'scenario':'phone_B_IP_restrict.xml','-i':'127.0.0.3','-p':'5060','-inf':INJECT_FILE_BRIDGE},
- {'scenario':'phone_B_IP_media_restrict.xml','-i':'127.0.0.3','-p':'5060','-inf':INJECT_FILE_BRIDGE},]
+ {'scenario':'phone_B_IP_media_restrict.xml','-i':'127.0.0.3','-p':'5060','-inf':INJECT_FILE_BRIDGE},
+ {'scenario':'phone_B_no_direction.xml','-i':'127.0.0.3','-p':'5060','-inf':INJECT_FILE_BRIDGE}]
self.passed = True
self.moh_start_events = 0
Added: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A_no_direction.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A_no_direction.xml?view=auto&rev=6483
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A_no_direction.xml (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A_no_direction.xml Mon Mar 2 22:13:14 2015
@@ -1,0 +1,87 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone A in hold scenario. This is the phone that initiates the call.">
+
+ <!-- Send the initial invite to start the call -->
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:[field2]@[remote_ip]:[remote_port]; SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ Max-Forwards: 70
+ From: "[field0]" <sip:[field0]@[local_ip]>;tag=[call_number]
+ To: <sip:[field2]@[remote_ip]>
+ Contact: sip:[field0]@[local_ip]:[local_port]
+ Call-ID: [call_id]
+ Cseq: [cseq] INVITE
+ User-Agent: AcmeCorp/2.0
+ Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
+ Supported: replaces, timer
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=root 238930722 238930722 IN IP[local_ip_type] [local_ip]
+ s=Asterisk PBX
+ c=IN IP[local_ip_type] [local_ip]
+ t=0 0
+ m=audio 15968 RTP/AVP 18 0 101
+ a=rtpmap:18 G729/8000
+ a=fmtp:18 annexb=no
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ a=fmtp:101 0-16
+ a=silenceSupp:off - - - -
+ a=ptime:20
+ a=sendrecv
+ ]]>
+ </send>
+
+ <!-- (Maybe) receive a Trying response from Phone B -->
+ <recv response="100" optional="true" />
+
+ <!-- (Maybe) receive a Ringing response from Phone B -->
+ <recv response="180" optional="true" />
+
+ <!-- Receive the OK response from Phone B -->
+ <recv response="200" />
+
+ <pause milliseconds="200"/>
+
+ <!-- Send ACK to Phone B -->
+ <send>
+ <![CDATA[
+ ACK sip:[field2]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ Max-Forwards: 70
+ From: "[field0]" <sip:[field0]@[local_ip]>;tag=[call_number]
+ To: <sip:[field2]@[remote_ip]>[peer_tag_param]
+ Contact: <sip:[field0]@[local_ip]:[local_port]>
+ Call-ID: [call_id]
+ Cseq: [cseq] ACK
+ User-Agent: AcmeCorp/2.0
+ Content-Length: [len]
+ ]]>
+ </send>
+
+ <!-- Wait some period of time while Phone B performs hold/unhold tasks -->
+ <pause milliseconds="3000"/>
+
+ <!-- Receive a BYE request from Phone B indicating scenario is over -->
+ <recv request="BYE"/>
+
+ <!-- Send OK response to Phone B -->
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ [last_Via:]
+ Content-Length: [len]
+ User-Agent: AcmeCorp/2.0
+ ]]>
+ </send>
+
+</scenario>
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A_no_direction.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A_no_direction.xml
------------------------------------------------------------------------------
svn:executable = *
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A_no_direction.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A_no_direction.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml?view=diff&rev=6483&r1=6482&r2=6483
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml (original)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml Mon Mar 2 22:13:14 2015
@@ -210,5 +210,6 @@
]]>
</send>
+ <recv response="200" />
</scenario>
Modified: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml?view=diff&rev=6483&r1=6482&r2=6483
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml (original)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml Mon Mar 2 22:13:14 2015
@@ -210,5 +210,6 @@
]]>
</send>
+ <recv response="200" />
</scenario>
Modified: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml?view=diff&rev=6483&r1=6482&r2=6483
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml (original)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml Mon Mar 2 22:13:14 2015
@@ -211,5 +211,6 @@
]]>
</send>
+ <recv response="200" />
</scenario>
Added: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_no_direction.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_no_direction.xml?view=auto&rev=6483
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_no_direction.xml (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_no_direction.xml Mon Mar 2 22:13:14 2015
@@ -1,0 +1,207 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone B in hold scenario. This is the phone that initiates the hold/unhold events.">
+
+ <!-- Receive the initial invite to initiate the call -->
+ <recv request="INVITE" crlf="true" />
+
+ <pause milliseconds="200"/>
+
+ <!-- Send a Trying response to Phone A -->
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ [last_Via:]
+ ]]>
+ </send>
+
+ <pause milliseconds="200"/>
+
+ <!-- Send a Ringing response to Phone A -->
+ <send>
+ <![CDATA[
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Server: AcmeCorp/2.0
+ Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
+ Supported: replaces, timer
+ Contact: <sip:[field2]@[local_ip]:[local_port]>
+ Content-Length: [len]
+ ]]>
+ </send>
+
+ <pause milliseconds="200"/>
+
+ <!-- Send an OK response to Phone A -->
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ [last_Via:]
+ Content-Type: application/sdp
+ Content-Length: [len]
+ User-Agent: AcmeCorp/2.0
+ Contact: sip:[field2]@[local_ip]:[local_port]
+ Allow: INVITE, ACK, CANCEL, BYE, INFO, UPDATE, OPTIONS, PRACK, REFER
+
+ v=0
+ o=- 1 1 IN IP[local_ip_type] [local_ip]
+ s=LO-CM
+ c=IN IP[local_ip_type] [local_ip]
+ t=0 0
+ m=audio 44064 RTP/AVP 0 101
+ a=rtpmap:0 PCMU/8000/1
+ a=rtpmap:101 telephone-event/8000/1
+ a=fmtp:101 0-15
+ ]]>
+ </send>
+
+ <!-- Receive ACK from Phone A -->
+ <recv request="ACK"/>
+
+ <!-- Wait some period of time, then send hold -->
+ <pause milliseconds="3000"/>
+
+ <!-- Send a re-invite to "Phone A" (Asterisk) to put the call on hold -->
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
+ From: sip:[field2]@[local_ip];tag=[call_number]
+ To: "[field0]" <sip:[field2]@[remote_ip]>
+ Call-ID: [call_id]
+ Cseq: [cseq] INVITE
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ Content-Type: application/sdp
+ Content-Length: [len]
+ Contact: sip:[field2]@[local_ip]:[local_port]
+ Max-Forwards: 70
+ User-Agent: AcmeCorp/2.0
+ Allow: INVITE, ACK, CANCEL, BYE, INFO, UPDATE, OPTIONS, PRACK, REFER
+
+ v=0
+ o=AcmeCorp 1393520501 1393520501 IN IP[local_ip_type] [local_ip]
+ s=AcmeCorp Call
+ c=IN IP4 0.0.0.0
+ t=0 0
+ a=inactive
+ m=audio 10000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000/1
+ a=rtpmap:101 telephone-event/8000/1
+ a=fmtp:101 0-15
+ ]]>
+ </send>
+
+ <!-- (Maybe) receive a Trying response from "Phone A" (Asterisk) -->
+ <recv response="100" optional="true" />
+
+ <!-- Receive the OK response from "Phone A" (Asterisk) -->
+ <recv response="200" />
+
+ <pause milliseconds="200"/>
+
+ <!-- Send ACK to "Phone A" (Asterisk) -->
+ <send>
+ <![CDATA[
+ ACK sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
+ From: sip:[field1]@[local_ip];tag=[call_number]
+ To: "[field0]" <sip:[field0]@[remote_ip]>[peer_tag_param]
+ Call-ID: [call_id]
+ Cseq: [cseq] ACK
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ Content-Length: [len]
+ Max-Forwards: 70
+ User-Agent: AcmeCorp/2.0
+ ]]>
+ </send>
+
+ <!-- Wait some period of time, then send the un-hold -->
+ <pause milliseconds="3000"/>
+
+ <!-- Send another re-invite to "Phone A" (Asterisk) to take the call out of hold;
+ omitting a=<direction> from SDP -->
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
+ From: sip:[field1]@[local_ip];tag=[call_number]
+ To: "[field0]" <sip:[field0]@[remote_ip]>[peer_tag_param]
+ Call-ID: [call_id]
+ Cseq: [cseq] INVITE
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ Content-Type: application/sdp
+ Content-Length: [len]
+ Contact: sip:[field2]@[local_ip]:[local_port]
+ Max-Forwards: 70
+ User-Agent: AcmeCorp/2.0
+ Allow: INVITE, ACK, CANCEL, BYE, INFO, UPDATE, OPTIONS, PRACK, REFER
+
+ v=0
+ o=AcmeCorp 7786706563 7786706563 IN IP[local_ip_type] [local_ip]
+ s=AcmeCorp Call
+ c=IN IP[local_ip_type] [local_ip]
+ t=0 0
+ m=audio 44170 RTP/AVP 0 101
+ a=rtpmap:0 PCMU/8000/1
+ a=rtpmap:101 telephone-event/8000/1
+ a=fmtp:101 0-15
+ ]]>
+ </send>
+
+ <!-- (Maybe) receive a Trying response from "Phone A" (Asterisk) -->
+ <recv response="100" optional="true" />
+
+ <!-- Receive the OK response from "Phone A" (Asterisk) -->
+ <recv response="200" />
+
+ <pause milliseconds="200"/>
+
+ <!-- Send ACK to "Phone A" (Asterisk) -->
+ <send>
+ <![CDATA[
+ ACK sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
+ From: sip:[field1]@[local_ip];tag=[call_number]
+ To: "[field0]" <sip:[field0]@[remote_ip]>[peer_tag_param]
+ Call-ID: [call_id]
+ Cseq: [cseq] ACK
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ Content-Length: [len]
+ Max-Forwards: 70
+ User-Agent: AcmeCorp/2.0
+ ]]>
+ </send>
+
+ <pause milliseconds="500"/>
+
+ <!-- Send a BYE request to Phone A indicating hold scenario is over -->
+ <send>
+ <![CDATA[
+ BYE sip:[field0]@[remote_ip]:[remote_port]; SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ Max-Forwards: 70
+ From: sip:[field1]@[local_ip];tag=[call_number]
+ To: "[field0]" <sip:[field0]@[remote_ip]>[peer_tag_param]
+ Contact: sip:[field1]@[local_ip]:[local_port]
+ Call-ID: [call_id]
+ Cseq: [cseq] BYE
+ User-Agent: AcmeCorp/2.0
+ X-Asterisk-HangupCause: Normal Clearing
+ X-Asterisk-HangupCauseCode: 16
+ Content-Length: [len]
+ ]]>
+ </send>
+
+ <!-- Receive the OK response from Phone A -->
+ <recv response="200" />
+
+</scenario>
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_no_direction.xml
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svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_no_direction.xml
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svn:executable = *
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_no_direction.xml
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svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_no_direction.xml
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svn:mime-type = text/plain
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