[asterisk-dev] [Code Review] SIP: more code refactoring, more unit tests

Russell Bryant russell at digium.com
Fri Feb 12 07:38:16 CST 2010


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This code looks good to me.  I'd like to see some invalid input test cases, too, though.


/trunk/channels/sip/reqresp_parser.c
<https://reviewboard.asterisk.org/r/499/#comment3379>

    What would you think about throwing some bogus input in here, too?


- Russell


On 2010-02-11 12:11:22, David Vossel wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/499/
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> 
> (Updated 2010-02-11 12:11:22)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> ----- Code Refactoring Changes ----
> - read_to_parts() moved to reqresp_parser.c and has been renamed as get_name_and_number()
> - get_in_brackets() moved to reqresp_parser.c
> - find_closing_quotes() added to sip_utils.h
> 
> ----- Logic Changes -----
> - get_name_and_number() now uses parse_uri() and get_calleridname() for parsing.  Before this change only names within quotes were found, when names not within quotes are possible.
> 
> ----- New Unit Tests -----
> sip_get_name_and_number_test
> sip_get_in_brackets_test
> 
> 
> Diffs
> -----
> 
>   /trunk/channels/chan_sip.c 246298 
>   /trunk/channels/sip/include/reqresp_parser.h 246298 
>   /trunk/channels/sip/include/sip_utils.h 246298 
>   /trunk/channels/sip/reqresp_parser.c 246298 
> 
> Diff: https://reviewboard.asterisk.org/r/499/diff
> 
> 
> Testing
> -------
> 
> -tested sip redirect to verify get_name_and_number still works as expected.
> -ran unit tests, verified they all passed.
> 
> 
> Thanks,
> 
> David
> 
>




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