[Asterisk-code-review] local: Add test for dialing with a removed/declined audio stream. (testsuite[18])
Joshua Colp
asteriskteam at digium.com
Fri Apr 30 04:07:08 CDT 2021
Joshua Colp has posted comments on this change. ( https://gerrit.asterisk.org/c/testsuite/+/15836 )
Change subject: local: Add test for dialing with a removed/declined audio stream.
......................................................................
Patch Set 1:
(1 comment)
https://gerrit.asterisk.org/c/testsuite/+/15836/1/tests/channels/local/local_removed_audio_stream_request/sipp/bob.xml
File tests/channels/local/local_removed_audio_stream_request/sipp/bob.xml:
https://gerrit.asterisk.org/c/testsuite/+/15836/1/tests/channels/local/local_removed_audio_stream_request/sipp/bob.xml@6
PS1, Line 6: <recv request="INVITE" crlf="true" />
> Use a regex to confirm that the SDP does not contain the declined stream.
I've adjusted the language for the test as receiving one stream in PJSIP land isn't really part of it, we're just testing that Local lets the call go through. The fact that only one stream appears on the PJSIP side is due to configuration of PJSIP itself.
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Gerrit-Project: testsuite
Gerrit-Branch: 18
Gerrit-Change-Id: If55c6ef39c322cb45f59e93bd4ef545957cf5c26
Gerrit-Change-Number: 15836
Gerrit-PatchSet: 1
Gerrit-Owner: Joshua Colp <jcolp at sangoma.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-Comment-Date: Fri, 30 Apr 2021 09:07:08 +0000
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Comment-In-Reply-To: Kevin Harwell <kharwell at digium.com>
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