[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
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1280/#review3825
-----------------------------------------------------------
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/
> -----------------------------------------------------------
>
> (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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20110706/bd1e653d/attachment-0001.htm>
More information about the asterisk-dev
mailing list