[asterisk-dev] [Code Review] 2808: Test Suite: nominal outgoing and two-party pjsip tests
Mark Michelson
reviewboard at asterisk.org
Fri Aug 30 14:53:23 CDT 2013
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Ship it!
Ship It!
- Mark Michelson
On Aug. 29, 2013, 9:29 p.m., jbigelow wrote:
>
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2808/
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>
> (Updated Aug. 29, 2013, 9:29 p.m.)
>
>
> Review request for Asterisk Developers and Mark Michelson.
>
>
> Bugs: ASTERISK-22283 and ASTERISK-22285
> https://issues.asterisk.org/jira/browse/ASTERISK-22283
> https://issues.asterisk.org/jira/browse/ASTERISK-22285
>
>
> Repository: testsuite
>
>
> Description
> -------
>
> These tests cover nominal outgoing and two-party PJSIP tests. Each test has iterations for IPv4/IPv6 and UDP/TCP. These cover tests 1 and 2 of the nominal outgoing tests and tests 1 and 2 of the nominal two-party call tests initiated by Alice as listed on the test plan page of the wiki.
>
> *NOTE:* Many of these tests currently fail due to multiple bugs in Asterisk which and have been reported. Changes may be required to the tests once the Asterisk bugs are resolved.
>
> Nominal outgoing tests:
> * playback - Originate a call from the uut to bob with directing the answered call to the Playback() application. Bob answers and listens for audio received from the uut. The uut then hangs up. This ensures that bob receives the audio from the uut.
>
> * echo - Originate a call from the uut to bob with directing the answered call to the Echo() application. Bob answers and sends audio while listening for audio received from the uut. Bob then hangs up. This ensures that bob receives the echoed audio from the uut.
>
> Nominal alice initiated two-party tests:
> * alice_hangs_up - Using three instances of Asterisk, Both alice and bob send audio to each other while both also listen for audio. Alice then initiates the hang. This ensures that both alice and bob receive audio.
>
> * bob_hangs_up - Using three instances of Asterisk, Both alice and bob send audio to each other while both also listen for audio. Bob then initiates the hang. This ensures that both alice and bob receive audio.
>
>
> Diffs
> -----
>
> /asterisk/trunk/tests/channels/pjsip/basic_calls/two_parties/tests.yaml PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/two_parties/nominal/tests.yaml PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/tests.yaml PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/bob_hangs_up/test-config.yaml PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/bob_hangs_up/configs/ast3/pjsip.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/bob_hangs_up/configs/ast3/extensions.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/bob_hangs_up/configs/ast2/pjsip.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/bob_hangs_up/configs/ast2/extensions.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/bob_hangs_up/configs/ast1/pjsip.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/bob_hangs_up/configs/ast1/extensions.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/alice_hangs_up/test-config.yaml PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/alice_hangs_up/configs/ast3/pjsip.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/alice_hangs_up/configs/ast3/extensions.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/alice_hangs_up/configs/ast2/pjsip.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/alice_hangs_up/configs/ast2/extensions.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/alice_hangs_up/configs/ast1/pjsip.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/alice_hangs_up/configs/ast1/extensions.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/tests.yaml 4098
> /asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/tests.yaml PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/tests.yaml PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/playback/test-config.yaml PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/playback/configs/ast2/pjsip.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/playback/configs/ast2/extensions.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/playback/configs/ast1/pjsip.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/playback/configs/ast1/extensions.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/echo/configs/ast1/extensions.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/echo/configs/ast1/pjsip.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/echo/configs/ast2/extensions.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/echo/configs/ast2/pjsip.conf PRE-CREATION
> /asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/echo/test-config.yaml PRE-CREATION
>
> Diff: https://reviewboard.asterisk.org/r/2808/diff/
>
>
> Testing
> -------
>
> Since many of these fail due to multiple bugs in Asterisk (see note above) I could only verify the tests based on limited successful behavior.
> * Verified that audio is received as specified for each test when calls actually go through
> * Verified failures are due to bugs found in Asterisk.
>
>
> Thanks,
>
> jbigelow
>
>
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