[asterisk-dev] [Code Review] TestSuite: Add tests for codec negotiation

mjordan reviewboard at asterisk.org
Wed Oct 12 13:29:33 CDT 2011


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1520/#review4502
-----------------------------------------------------------



asterisk/trunk/tests/channels/SIP/codec_negotiation/run-test
<https://reviewboard.asterisk.org/r/1520/#comment8678>

    Nuts.  I hadn't tackled making SIPpTest inherit from TestCase, but at this juncture, it'd sure be nice if it did (so that it got all the other bells and whistles stuck in there).
    
    Not a problem if you don't want to tackle that on this issue, but its something we should probably look into.


- mjordan


On Oct. 12, 2011, 1:09 p.m., opticron wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1520/
> -----------------------------------------------------------
> 
> (Updated Oct. 12, 2011, 1:09 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> Add codec and stream negotiation tests for varying single stream, incompatible stream, and multistream situations. This requires the changes in https://reviewboard.asterisk.org/r/1516/ to pass since it corrects several issues with how Asterisk deals with SDP.
> 
> 
> Diffs
> -----
> 
>   asterisk/trunk/tests/channels/SIP/codec_negotiation/configs/ast1/extensions.conf PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/codec_negotiation/configs/ast1/sip.conf PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/codec_negotiation/run-test PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/avt_streams.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_audio.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_text.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_video.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/multistream.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/orderstream.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_audio.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_image.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_image_inverse.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_text.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_text_inverse.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_video.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_video_inverse.xml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/codec_negotiation/test-config.yaml PRE-CREATION 
>   asterisk/trunk/tests/channels/SIP/tests.yaml 2515 
> 
> Diff: https://reviewboard.asterisk.org/r/1520/diff
> 
> 
> Testing
> -------
> 
> 
> Thanks,
> 
> opticron
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20111012/6fb2ba9d/attachment.htm>


More information about the asterisk-dev mailing list