[asterisk-dev] [Code Review] 'directrtpsetup' option test

Joshua Colp reviewboard at asterisk.org
Sat Nov 17 10:23:18 CST 2012


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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2185/
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(Updated Nov. 17, 2012, 10:23 a.m.)


Review request for Asterisk Developers.


Changes
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Incorporated feedback.


Summary
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This test covers the 'directrtpsetup' option present within chan_sip. It creates a call going through Asterisk and examines the SDP to ensure that the address for media is of the caller, and not that of Asterisk. It also examines the answer SDP to confirm that the address for media is that of the called party.


Diffs (updated)
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  /asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/extensions.conf PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/sip.conf PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_A.xml PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_B.xml PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/directrtpsetup/test-config.yaml PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/tests.yaml 3517 

Diff: https://reviewboard.asterisk.org/r/2185/diff


Testing
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Ran test to confirm it works, purposely broke test to confirm it fails. Ate some toast. It was crunchy.


Thanks,

Joshua

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