Author: wdoekes Date: Sun Oct 12 03:24:59 2014 New Revision: 5715 URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=5715 Log: chan_sip: Test unscheduling reINVITE after call hangup.
Belongs with chan_sip commit r425296 (1.8). Tests that no reINVITE is sent after dialog hangup. Review: https://reviewboard.asterisk.org/r/4055/ Added: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/ asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/ asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/extensions.conf (with props) asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/sip.conf (with props) asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/ asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/alice.xml (with props) asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/bob.xml (with props) asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml (with props) Modified: asterisk/trunk/tests/channels/SIP/tests.yaml Added: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/extensions.conf URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/extensions.conf?view=auto&rev=5715 ============================================================================== --- asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/extensions.conf (added) +++ asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/extensions.conf Sun Oct 12 03:24:59 2014 @@ -1,0 +1,2 @@ +[default] +exten => bob,1,Dial(SIP/bob) Propchange: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/extensions.conf ------------------------------------------------------------------------------ svn:eol-style = native Propchange: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/extensions.conf ------------------------------------------------------------------------------ svn:keywords = Author Date Id Revision Propchange: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/extensions.conf ------------------------------------------------------------------------------ svn:mime-type = text/plain Added: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/sip.conf URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/sip.conf?view=auto&rev=5715 ============================================================================== --- asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/sip.conf (added) +++ asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/sip.conf Sun Oct 12 03:24:59 2014 @@ -1,0 +1,21 @@ +[general] +udpbindaddr=127.0.0.1:5060 +; debugging is nice +sipdebug=yes +; we require/expect directmedia reinvites for our test +directmedia=yes +; wait 3.2 seconds to timeout retransmitting 200 to re-invite +; instead of 32 seconds +timert1=50 +; allow fax +t38pt_udptl=yes + +[alice] +host=127.0.0.1 +port=5062 +type=friend + +[bob] +host=127.0.0.1 +port=5063 +type=friend Propchange: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/sip.conf ------------------------------------------------------------------------------ svn:eol-style = native Propchange: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/sip.conf ------------------------------------------------------------------------------ svn:keywords = Author Date Id Revision Propchange: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/sip.conf ------------------------------------------------------------------------------ svn:mime-type = text/plain Added: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/alice.xml URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/alice.xml?view=auto&rev=5715 ============================================================================== --- asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/alice.xml (added) +++ asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/alice.xml Sun Oct 12 03:24:59 2014 @@ -1,0 +1,98 @@ +<?xml version="1.0" encoding="ISO-8859-1"?> +<!DOCTYPE scenario SYSTEM "sipp.dtd"> +<!-- Walter Doekes, 2014 for asterisk bug ASTERISK-22791 --> +<scenario name="ASTERISK-22791-alice"> + + <send retrans="500"> + <![CDATA[ + + INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 + Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] + From: alice <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number] + To: [service] <sip:[service]@[remote_ip]:[remote_port]> + Call-ID: [call_id] + CSeq: 1 INVITE + Contact: <sip:alice@[local_ip]:[local_port]> + Content-Type: application/sdp + Content-Length: [len] + + v=0 + o=user1 [pid][call_number][cseq] [pid][call_number][cseq] 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 8 + a=rtpmap:0 PCMU/8000 + + ]]> + </send> + + <recv response="100" optional="true"> + </recv> + + <recv response="180" optional="true"> + </recv> + + <recv response="200" rrs="true"> + </recv> + + <send> + <![CDATA[ + + ACK [next_url] SIP/2.0 + Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] + [last_From:] + [last_To:] + [last_Call-ID:] + CSeq: 1 ACK + Contact: <sip:alice@[local_ip]:[local_port]> + Content-Length: 0 + + ]]> + </send> + + <!-- expect directmedia reinvite --> + <recv request="INVITE"> + </recv> + + <!-- reject it, because we were going to hang up --> + <send retrans="500"> + <![CDATA[ + + SIP/2.0 491 Request Pending + [last_Via:] + [last_From:] + [last_To:];tag=[pid]SIPpTag01[call_number] + [last_Call-ID:] + [last_CSeq:] + Contact: <sip:alice@[local_ip]:[local_port]> + Content-Length: 0 + + ]]> + </send> + + <recv request="ACK"> + </recv> + + <!-- done with the call --> + <send retrans="500"> + <![CDATA[ + + BYE [next_url] SIP/2.0 + Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] + From: alice <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number] + To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] + [last_Call-ID:] + CSeq: 2 BYE + Content-Length: 0 + + ]]> + </send> + + <recv response="200"> + </recv> + + <!-- at this point we *don't* want to see another reINVITE --> + <timewait milliseconds="3000"/> + +</scenario> Propchange: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/alice.xml ------------------------------------------------------------------------------ svn:eol-style = native Propchange: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/alice.xml ------------------------------------------------------------------------------ svn:keywords = Author Date Id Revision Propchange: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/alice.xml ------------------------------------------------------------------------------ svn:mime-type = text/plain Added: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/bob.xml URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/bob.xml?view=auto&rev=5715 ============================================================================== --- asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/bob.xml (added) +++ asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/bob.xml Sun Oct 12 03:24:59 2014 @@ -1,0 +1,110 @@ +<?xml version="1.0" encoding="ISO-8859-1"?> +<!DOCTYPE scenario SYSTEM "sipp.dtd"> +<!-- Walter Doekes, 2014 for asterisk bug ASTERISK-22791 --> +<scenario name="ASTERISK-22791-bob"> + + <!-- expect call from alice --> + <recv request="INVITE"> + </recv> + + <send> + <![CDATA[ + + SIP/2.0 180 Ringing + [last_Via:] + [last_From:] + [last_To:];tag=[pid]SIPpTag01[call_number] + [last_Call-ID:] + [last_CSeq:] + Contact: <sip:bob@[local_ip]:[local_port]> + Content-Length: 0 + + ]]> + </send> + + <send retrans="500"> + <![CDATA[ + + SIP/2.0 200 OK + [last_Via:] + [last_From:] + [last_To:];tag=[pid]SIPpTag01[call_number] + [last_Call-ID:] + [last_CSeq:] + Contact: <sip:bob@[local_ip]:[local_port]> + Content-Type: application/sdp + Content-Length: [len] + + v=0 + o=user1 [pid][call_number][cseq] [pid][call_number][cseq] 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:0 PCMU/8000 + + ]]> + </send> + + <recv request="ACK"> + </recv> + + <!-- expect directmedia reinvite --> + <recv request="INVITE"> + </recv> + + <label id="reinvite"/> + + <!-- fine --> + <send retrans="500"> + <![CDATA[ + + SIP/2.0 200 OK + [last_Via:] + [last_From:] + [last_To:];tag=[pid]SIPpTag01[call_number] + [last_Call-ID:] + [last_CSeq:] + Contact: <sip:bob@[local_ip]:[local_port]> + Content-Type: application/sdp + Content-Length: [len] + + v=0 + o=user1 [pid][call_number][cseq] [pid][call_number][cseq] 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:0 PCMU/8000 + + ]]> + </send> + + <recv request="ACK"> + </recv> + + <recv request="INVITE" optional="true" next="reinvite"> + </recv> + + <!-- the call gets hung up --> + <recv request="BYE"> + </recv> + + <send> + <![CDATA[ + + SIP/2.0 200 OK + [last_Via:] + [last_From:] + [last_To:] + [last_Call-ID:] + [last_CSeq:] + Content-Length: 0 + + ]]> + </send> + + <!-- wait a bit to be able to retransmit our 200 --> + <timewait milliseconds="3000"/> + +</scenario> Propchange: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/bob.xml ------------------------------------------------------------------------------ svn:eol-style = native Propchange: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/bob.xml ------------------------------------------------------------------------------ svn:keywords = Author Date Id Revision Propchange: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/bob.xml ------------------------------------------------------------------------------ svn:mime-type = text/plain Added: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml?view=auto&rev=5715 ============================================================================== --- asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml (added) +++ asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml Sun Oct 12 03:24:59 2014 @@ -1,0 +1,36 @@ +testinfo: + summary: 'Test that we get no reINVITE after 491 after BYE' + description: | + 'This tests a scenario where asterisk initiates a reINVITE -- + which gets 491d -- and the call is hung up in the mean time. + If the bug is not fixed, we get another reINVITE (with + reversed From and To headers). See bug: ASTERISK-22791' + +properties: + minversion: '1.8.32.0' + dependencies: + - python: 'starpy' + - sipp: + version: 'v3.1' + - asterisk : 'chan_sip' + tags: + - SIP + +test-modules: + test-object: + config-section: sipp-config + typename: 'sipp.SIPpTestCase' + +sipp-config: + reactor-timeout: 20 + fail-on-any: True + test-iterations: + - + scenarios: + - {'key-args': {'scenario': 'bob.xml', + '-p': '5063', + '-default_behaviors': '-bye'}} + - {'key-args': {'scenario': 'alice.xml', + '-p': '5062', + '-s': 'bob', + '-default_behaviors': '-bye,abortunexp'}} Propchange: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml ------------------------------------------------------------------------------ svn:eol-style = native Propchange: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml ------------------------------------------------------------------------------ svn:keywords = Author Date Id Revision Propchange: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml ------------------------------------------------------------------------------ svn:mime-type = text/plain Modified: asterisk/trunk/tests/channels/SIP/tests.yaml URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/tests.yaml?view=diff&rev=5715&r1=5714&r2=5715 ============================================================================== --- asterisk/trunk/tests/channels/SIP/tests.yaml (original) +++ asterisk/trunk/tests/channels/SIP/tests.yaml Sun Oct 12 03:24:59 2014 @@ -71,3 +71,4 @@ - dir: 'ami' - test: 'invite_retransmit' - test: 'no_ack_dialog_cleanup' + - test: 'no_reinvite_after_491' -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- svn-commits mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/svn-commits
