[Asterisk-code-review] channels/pjsip/basic calls/outgoing/nominal/early media: tes... (testsuite[master])

[Richard Mudgett]

Richard Mudgett has posted comments on this change. ( https://gerrit.asterisk.org/8303 )

Change subject: channels/pjsip/basic_calls/outgoing/nominal/early_media: test forked early media
......................................................................


Patch Set 1: Code-Review-1

(6 comments)

https://gerrit.asterisk.org/#/c/8303/1/tests/channels/pjsip/basic_calls/outgoing/nominal/early_media/configs/ast1/extensions.conf
File tests/channels/pjsip/basic_calls/outgoing/nominal/early_media/configs/ast1/extensions.conf:

https://gerrit.asterisk.org/#/c/8303/1/tests/channels/pjsip/basic_calls/outgoing/nominal/early_media/configs/ast1/extensions.conf@10
PS1, Line 10: exten => s,1,UserEvent(rtpinfo,dest: ${CHANNEL(rtp,dest)})
This dialplan routine must return.
Before returning you should make the call hangup:

same = n,SoftHangup(PJSIP/ua1)
same = n,Return()


https://gerrit.asterisk.org/#/c/8303/1/tests/channels/pjsip/basic_calls/outgoing/nominal/early_media/sipp/ua1_invite_recv.xml
File tests/channels/pjsip/basic_calls/outgoing/nominal/early_media/sipp/ua1_invite_recv.xml:

https://gerrit.asterisk.org/#/c/8303/1/tests/channels/pjsip/basic_calls/outgoing/nominal/early_media/sipp/ua1_invite_recv.xml@16
PS1, Line 16:   
red blob
i.e. Trailing whitespace


https://gerrit.asterisk.org/#/c/8303/1/tests/channels/pjsip/basic_calls/outgoing/nominal/early_media/sipp/ua1_invite_recv.xml@40
PS1, Line 40:   
red blob


https://gerrit.asterisk.org/#/c/8303/1/tests/channels/pjsip/basic_calls/outgoing/nominal/early_media/sipp/ua1_invite_recv.xml@72
PS1, Line 72:   <send retrans="500">
            :     <![CDATA[
            : 
            :       BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
            :       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
            :       From: ua1 <sip:ua1@[local_ip]:[local_port]>;tag=[pid]SIPpForkTag01[call_number]
            :       To: [$remote_tag]
            :       [last_Call-ID:]
            :       CSeq: [cseq] BYE
            :       Contact: <sip:[service]@[local_ip]:[local_port];transport=[transport]>
            :       Max-Forwards: 70
            :       Subject: Path Test
            :       Content-Length: 0
            : 
            :     ]]>
            :   </send>
This test is fragile because of a race condition.  The sipp script may or may not hangup the channel before Asterisk has a chance to execute the on-answer dialplan routine.

You need to make it so Asterisk hangs up the call instead.  Thus the sipp scenario needs to receive the BYE instead of send it.


https://gerrit.asterisk.org/#/c/8303/1/tests/channels/pjsip/basic_calls/outgoing/nominal/early_media/test-config.yaml
File tests/channels/pjsip/basic_calls/outgoing/nominal/early_media/test-config.yaml:

https://gerrit.asterisk.org/#/c/8303/1/tests/channels/pjsip/basic_calls/outgoing/nominal/early_media/test-config.yaml@12
PS1, Line 12:     minversion: '14.0'
Minversion needs to be 15.4.0


https://gerrit.asterisk.org/#/c/8303/1/tests/channels/pjsip/basic_calls/outgoing/nominal/early_media/test-config.yaml@17
PS1, Line 17:         - asterisk : 'res_pjsip'
Add test dependencies:
- asterisk: 'app_dial'
- asterisk: 'app_softhangup'
- asterisk: 'app_stack'
- asterisk: 'app_userevent'
- asterisk: 'func_channel'



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Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-MessageType: comment
Gerrit-Change-Id: I62accd4f993a6f42c77661e4943c1e9df5fbe04c
Gerrit-Change-Number: 8303
Gerrit-PatchSet: 1
Gerrit-Owner: lvl <digium at lvlconsultancy.nl>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
Gerrit-Comment-Date: Tue, 06 Mar 2018 20:38:03 +0000
Gerrit-HasComments: Yes
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