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

[rmudgett]

> On July 6, 2011, 12:37 p.m., Paul Belanger wrote:
> > /asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast1/extensions.conf, lines 10-11
> > <https://reviewboard.asterisk.org/r/1280/diff/2/?file=17222#file17222line10>
> >
> >     Remove

I would rather leave this comment commented out dialplan code.  There is an associated comment with commented out code in the run-test script about not dropping the AGI connection.  If that code is uncommented then this dialplan code would be needed.

These comments are more about documenting what should not be done to get a more reliable test.


> On July 6, 2011, 12:37 p.m., Paul Belanger wrote:
> > /asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast1/sip.conf, lines 7-8
> > <https://reviewboard.asterisk.org/r/1280/diff/2/?file=17223#file17223line7>
> >
> >     same, can be removed

done


> On July 6, 2011, 12:37 p.m., Paul Belanger wrote:
> > /asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast2/sip.conf, lines 7-8
> > <https://reviewboard.asterisk.org/r/1280/diff/2/?file=17225#file17225line7>
> >
> >     remove

done


> On July 6, 2011, 12:37 p.m., Paul Belanger wrote:
> > /asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/sip.conf, lines 7-8
> > <https://reviewboard.asterisk.org/r/1280/diff/2/?file=17231#file17231line7>
> >
> >     same, remove

done


> On July 6, 2011, 12:37 p.m., Paul Belanger wrote:
> > /asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast2/sip.conf, line 5
> > <https://reviewboard.asterisk.org/r/1280/diff/2/?file=17225#file17225line5>
> >
> >     Rather then changing the port, increment the loopback adapter.
> >     
> >     udpbindaddr=127.0.0.2:5060

Is there some benefit to using a different IP address rather than attaching to a different port?
Using a different port should cause different code paths to be executed.


> On July 6, 2011, 12:37 p.m., Paul Belanger wrote:
> > /asterisk/trunk/tests/channels/SIP/secure_bridge_media/run-test, line 42
> > <https://reviewboard.asterisk.org/r/1280/diff/2/?file=17241#file17241line42>
> >
> >     convert this to originate, console requires a soundcard and not all remote agents have them.

This test is about using Set(CHANNEL(secure_bridge_media)=1) to initiate a SIP SRTP call.

I suppose using originate to a local channel might be possible.  Not sure.


> On July 6, 2011, 12:37 p.m., Paul Belanger wrote:
> > /asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast1/sip.conf, lines 37-49
> > <https://reviewboard.asterisk.org/r/1280/diff/2/?file=17223#file17223line37>
> >
> >     I'd like to see 'secret' added, more coverage.

ok


- rmudgett


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On June 24, 2011, 12:41 p.m., rmudgett wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1280/
> -----------------------------------------------------------
> 
> (Updated June 24, 2011, 12:41 p.m.)
> 
> 
> Review request for Asterisk Developers, Paul Belanger and jrose.
> 
> 
> Summary
> -------
> 
> These tests check the ability to get SRTP connections:
> 1) sip_srtp test establishes a SIP call with SRTP to see if the call can get connected.
> 2) noload_res_srtp test checks to see if a normal SIP call can still be done when SRTP is not loaded.
> 3) noload_res_srtp_attemtp_srtp test checks to see if the call will fail if SRTP is not enabled and an incoming call requests it.
> 4) secure_bridge_media test checks to see if SRTP can be requested dynamically for an outgoing call.
> 
> 
> Diffs
> -----
> 
>   /asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast2/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast2/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/modules.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/noload_res_srtp/run-test PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/noload_res_srtp/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/manager.general.conf.inc PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast2/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast2/modules.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast2/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/run-test PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/secure_bridge_media/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/secure_bridge_media/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/secure_bridge_media/configs/ast2/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/secure_bridge_media/configs/ast2/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/secure_bridge_media/run-test PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/secure_bridge_media/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_srtp/configs/ast1/extensions.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/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 1679 
> 
> Diff: https://reviewboard.asterisk.org/r/1280/diff
> 
> 
> Testing
> -------
> 
> The tests pass and the debug output shows what is expected for each test.
> 
> 
> Thanks,
> 
> rmudgett
> 
>

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