[asterisk-bugs] [JIRA] (ASTERISK-22217) TestSuite sip_hold test fails in SIPp scenarios on unexpected SIP INVITE requests
Jonathan Rose (JIRA)
noreply at issues.asterisk.org
Thu Aug 8 14:41:03 CDT 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-22217?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=208628#comment-208628 ]
Jonathan Rose commented on ASTERISK-22217:
------------------------------------------
First part of the failure appears to be resolvable by setting directmedia=no in configs.
This exposes another failure though. Asterisk 12 splits MusicOnHold events to MusicOnHoldStart and MusicOnHoldStop so that only the channel snapshot is exposed as data rather than the event name. This does not appear to be taken into account for the test.
> TestSuite sip_hold test fails in SIPp scenarios on unexpected SIP INVITE requests
> ---------------------------------------------------------------------------------
>
> Key: ASTERISK-22217
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-22217
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/General, Tests/testsuite
> Affects Versions: 12
> Reporter: Matt Jordan
> Labels: Asterisk12
>
> It appears as if some re-INVITE messages are occurring - apparently to force directmedia - that weren't before. Yay.
> {noformat}
> Running ['tests/channels/SIP/sip_hold/run-test'] ...
> [Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:378 processEnded: Resolving remote host '127.0.0.1'...
> [Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:378 processEnded: Done.
> [Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:378 processEnded: 2013-07-30 13:11:14:986 1375204274.986483: Aborting call on unexpected message for Call-Id '37f12a637c8d7f1b17769da73ecf6c75 at 127.0.0.1:5060': while pausing (index 6), received 'INVITE sip:phone_B at 127.0.0.3:5060;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK278ebec0
> Max-Forwards: 70
> From: "phone_A" <sip:phone_A at 127.0.0.1>;tag=as50c16487
> To: <sip:127.0.0.3>;tag=1
> Contact: <sip:phone_A at 127.0.0.1:5060>
> Call-ID: 37f12a637c8d7f1b17769da73ecf6c75 at 127.0.0.1:5060
> CSeq: 103 INVITE
> User-Agent: Asterisk PBX SVN-trunk-r395686M
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> X-asterisk-Info: SIP re-invite (External RTP bridge)
> Content-Type: application/sdp
> Content-Length: 267
> v=0
> o=root 1387452781 1387452782 IN IP4 127.0.0.2
> s=Asterisk PBX SVN-trunk-r395686M
> c=IN IP4 127.0.0.2
> t=0 0
> m=audio 2226 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> '.
> [Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:467 __scenario_callback: SIPp Scenario phone_B_media_restrict.xml Failed [1]
> [Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:475 __evaluate_scenario_results: SIPp Scenario phone_B_media_restrict.xml Failed
> [Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:378 processEnded: Resolving remote host '127.0.0.1'...
> [Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:378 processEnded: Done.
> [Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:378 processEnded: 2013-07-30 13:11:14:988 1375204274.988396: Aborting call on unexpected message for Call-Id '1-7543 at 127.0.0.2': while expecting 'BYE' (index 6), received 'INVITE sip:phone_A at 127.0.0.2:5060 SIP/2.0
> Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0c48a3e2
> Max-Forwards: 70
> From: <sip:basicdial at 127.0.0.1:5060;user=phone>;tag=as004df771
> To: phone_A <sip:phone_A at 127.0.0.2:5060>;tag=1
> Contact: <sip:basicdial at 127.0.0.1:5060>
> Call-ID: 1-7543 at 127.0.0.2
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX SVN-trunk-r395686M
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> X-asterisk-Info: SIP re-invite (External RTP bridge)
> Content-Type: application/sdp
> Content-Length: 268
> v=0
> o=root 1242229468 1242229469 IN IP4 127.0.0.1
> s=Asterisk PBX SVN-trunk-r395686M
> c=IN IP4 127.0.0.1
> t=0 0
> m=audio 10848 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> '.
> [Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:467 __scenario_callback: SIPp Scenario phone_A.xml Failed [1]
> [Jul 30 13:11:14] WARNING[5035]: asterisk.sipp:475 __evaluate_scenario_results: SIPp Scenario phone_A.xml Failed
> [Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:378 processEnded: Resolving remote host '127.0.0.1'...
> [Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:378 processEnded: Done.
> [Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:378 processEnded: 2013-07-30 13:11:15:193 1375204275.193873: Aborting call on unexpected message for Call-Id '1-7548 at 127.0.0.2': while expecting '180' (index 2), received 'SIP/2.0 603 Declined
> Via: SIP/2.0/UDP 127.0.0.2:5060;branch=z9hG4bK-7548-1-0;received=127.0.0.2
> From: phone_A <sip:phone_A at 127.0.0.2:5060>;tag=1
> To: <sip:basicdial at 127.0.0.1:5060;user=phone>;tag=as4a10f805
> Call-ID: 1-7548 at 127.0.0.2
> CSeq: 1 INVITE
> Server: Asterisk PBX SVN-trunk-r395686M
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> Content-Length: 0
> '.
> [Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:467 __scenario_callback: SIPp Scenario phone_A.xml Failed [1]
> [Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:475 __evaluate_scenario_results: SIPp Scenario phone_A.xml Failed
> [Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:378 processEnded: Resolving remote host '127.0.0.1'... Done.
> [Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:378 processEnded: 2013-07-30 13:11:15:688 1375204275.688405: Aborting call on unexpected message for Call-Id '37f12a637c8d7f1b17769da73ecf6c75 at 127.0.0.1:5060': while pausing (index 6), received 'INVITE sip:phone_B at 127.0.0.3:5060;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK7afe8fbf
> Max-Forwards: 70
> From: "phone_A" <sip:phone_A at 127.0.0.1>;tag=as50c16487
> To: <sip:127.0.0.3>;tag=1
> Contact: <sip:phone_A at 127.0.0.1:5060>
> Call-ID: 37f12a637c8d7f1b17769da73ecf6c75 at 127.0.0.1:5060
> CSeq: 104 INVITE
> User-Agent: Asterisk PBX SVN-trunk-r395686M
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> X-asterisk-Info: SIP re-invite (External RTP bridge)
> Content-Type: application/sdp
> Content-Length: 267
> v=0
> o=root 1387452781 1387452783 IN IP4 127.0.0.2
> s=Asterisk PBX SVN-trunk-r395686M
> c=IN IP4 127.0.0.2
> t=0 0
> m=audio 2226 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> '.
> [Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:467 __scenario_callback: SIPp Scenario phone_B_IP_restrict.xml Failed [1]
> [Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:475 __evaluate_scenario_results: SIPp Scenario phone_B_IP_restrict.xml Failed
> [Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:378 processEnded: Resolving remote host '127.0.0.1'... Done.
> [Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:378 processEnded: 2013-07-30 13:11:15:803 1375204275.803695: Aborting call on unexpected message for Call-Id '1-8043 at 127.0.0.2': while expecting '100' (index 1), received 'SIP/2.0 503 Unavailable
> Via: SIP/2.0/UDP 127.0.0.2:5060;branch=z9hG4bK-8043-1-0;received=127.0.0.2
> From: phone_A <sip:phone_A at 127.0.0.2:5060>;tag=1
> To: <sip:basicdial at 127.0.0.1:5060;user=phone>;tag=as44182b0f
> Call-ID: 1-8043 at 127.0.0.2
> CSeq: 1 INVITE
> Server: Asterisk PBX SVN-trunk-r395686M
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> Content-Length: 0
> '.
> [Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:467 __scenario_callback: SIPp Scenario phone_A.xml Failed [1]
> [Jul 30 13:11:15] WARNING[5035]: asterisk.sipp:475 __evaluate_scenario_results: SIPp Scenario phone_A.xml Failed
> [Jul 30 13:11:19] WARNING[5035]: asterisk.sipp:378 processEnded: Resolving remote host '127.0.0.1'... Done.
> [Jul 30 13:11:19] WARNING[5035]: asterisk.sipp:378 processEnded: 2013-07-30 13:11:19:599 1375204279.599578: Aborting call on unexpected message for Call-Id '37f12a637c8d7f1b17769da73ecf6c75 at 127.0.0.1:5060': while pausing (index 12), received 'INVITE sip:phone_B at 127.0.0.3:5060;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK66e8c55e
> Max-Forwards: 70
> From: phone_A <sip:phone_B at 127.0.0.1>;tag=as50c16487
> To: phone_B <sip:phone_B at 127.0.0.3:5060>;tag=1
> Contact: <sip:phone_A at 127.0.0.1:5060>
> Call-ID: 37f12a637c8d7f1b17769da73ecf6c75 at 127.0.0.1:5060
> CSeq: 105 INVITE
> User-Agent: Asterisk PBX SVN-trunk-r395686M
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> X-asterisk-Info: SIP re-invite (External RTP bridge)
> Content-Type: application/sdp
> Content-Length: 268
> v=0
> o=root 1387452781 1387452785 IN IP4 127.0.0.1
> s=Asterisk PBX SVN-trunk-r395686M
> c=IN IP4 127.0.0.1
> t=0 0
> m=audio 12068 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=recvonly
> '.
> [Jul 30 13:11:19] WARNING[5035]: asterisk.sipp:467 __scenario_callback: SIPp Scenario phone_B_IP_media_restrict.xml Failed [1]
> [Jul 30 13:11:19] WARNING[5035]: asterisk.sipp:475 __evaluate_scenario_results: SIPp Scenario phone_B_IP_media_restrict.xml Failed
> [Jul 30 13:11:20] ERROR[5035]: __main__:116 main: Failed to receive 6 MOH start events (received 0)
> [Jul 30 13:11:20] ERROR[5035]: __main__:119 main: Failed to receive 6 MOH stop events (received 0)
> [Jul 30 13:11:20] ERROR[5035]: __main__:122 main: Failed to receive 6 user test events (received 4)
> {noformat}
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