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

Paul Belanger reviewboard at asterisk.org
Wed Jul 6 12:37:55 CDT 2011


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This looks good, a few minor comments.


/asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast1/extensions.conf
<https://reviewboard.asterisk.org/r/1280/#comment7633>

    Remove



/asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast1/sip.conf
<https://reviewboard.asterisk.org/r/1280/#comment7634>

    same, can be removed



/asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast1/sip.conf
<https://reviewboard.asterisk.org/r/1280/#comment7635>

    I'd like to see 'secret' added, more coverage.



/asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast2/extensions.conf
<https://reviewboard.asterisk.org/r/1280/#comment7636>

    same



/asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast2/sip.conf
<https://reviewboard.asterisk.org/r/1280/#comment7637>

    Rather then changing the port, increment the loopback adapter.
    
    udpbindaddr=127.0.0.2:5060



/asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast2/sip.conf
<https://reviewboard.asterisk.org/r/1280/#comment7638>

    remove



/asterisk/trunk/tests/channels/SIP/noload_res_srtp/configs/ast2/sip.conf
<https://reviewboard.asterisk.org/r/1280/#comment7639>

    same comment about 'secret'



/asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/extensions.conf
<https://reviewboard.asterisk.org/r/1280/#comment7640>

    same



/asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/sip.conf
<https://reviewboard.asterisk.org/r/1280/#comment7641>

    same, remove



/asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast2/extensions.conf
<https://reviewboard.asterisk.org/r/1280/#comment7642>

    same



/asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast2/sip.conf
<https://reviewboard.asterisk.org/r/1280/#comment7643>

    increment loopback



/asterisk/trunk/tests/channels/SIP/secure_bridge_media/run-test
<https://reviewboard.asterisk.org/r/1280/#comment7644>

    convert this to originate, console requires a soundcard and not all remote agents have them.


- Paul


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/
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> 
> (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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