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

[Joshua Colp]

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https://reviewboard.asterisk.org/r/2185/
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Review request for Asterisk Developers.


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
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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/run-test 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 3508 

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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