[asterisk-dev] [Code Review] sip_srtp test added to external test suite

[rmudgett]

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https://reviewboard.asterisk.org/r/1280/
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Review request for Asterisk Developers, Paul Belanger and jrose.


Summary
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This test establishes a SIP call with SRTP to see if the call can get connected.


Diffs
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  /asterisk/trunk/tests/channels/SIP/sip_srtp/configs/ast1/extensions.conf PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_srtp/configs/ast1/manager.conf PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_srtp/configs/ast1/sip.conf PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_srtp/configs/ast2/extensions.conf PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_srtp/configs/ast2/manager.conf PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_srtp/configs/ast2/sip.conf PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_srtp/run-test PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/sip_srtp/test-config.yaml PRE-CREATION 
  /asterisk/trunk/tests/channels/SIP/tests.yaml 1668 

Diff: https://reviewboard.asterisk.org/r/1280/diff


Testing
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The test passes and the debug output shows that the call does get connected with SRTP.


Thanks,

rmudgett

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