[asterisk-dev] [Code Review] 2683: Add DTLS-SRTP support to chan_pjsip

[Joshua Colp]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2683/#review9156
-----------------------------------------------------------



trunk/res/res_sip_sdp_rtp.c
<https://reviewboard.asterisk.org/r/2683/#comment18059>

    You can use pj_stricmp2 and not duplicate the string.



trunk/res/res_sip_sdp_rtp.c
<https://reviewboard.asterisk.org/r/2683/#comment18061>

    You can probably get away with:
    
    char fingerprint[value->slen];
    
    ast_copy_pj_str(fingerprint, &attr->value, sizeof(fingerprint));



trunk/res/res_sip_sdp_rtp.c
<https://reviewboard.asterisk.org/r/2683/#comment18062>

    Minor comment for the below uses of pj_cstr.
    
    The trend in pjsip, and in the resulting chan_pjsip code, has been to just statically declare these unchanging pj_strs. I'll leave it up to your discretion.


- Joshua Colp


On July 18, 2013, 1:40 p.m., opticron wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2683/
> -----------------------------------------------------------
> 
> (Updated July 18, 2013, 1:40 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-21419
>     https://issues.asterisk.org/jira/browse/ASTERISK-21419
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> This patch introduces DTLS-SRTP support to chan_pjsip and the options necessary to configure it including an option to allow choosing between 32 and 80 byte SRTP tag lengths.
> 
> During the implementation and testing of this patch, three other bugs were found and their fixes are included with this patch. The two in chan_sip were a segfault relating to DTLS setup and mistaken call rejection.  The third bug fix prevents chan_pjsip from attempting to perform bridge optimization between two endpoints if either of them is running any form of SRTP.
> 
> 
> Diffs
> -----
> 
>   trunk/channels/chan_gulp.c 394643 
>   trunk/channels/chan_sip.c 394643 
>   trunk/include/asterisk/res_sip.h 394643 
>   trunk/include/asterisk/res_sip_session.h 394643 
>   trunk/res/res_sip.c 394643 
>   trunk/res/res_sip/sip_configuration.c 394643 
>   trunk/res/res_sip_sdp_rtp.c 394643 
>   trunk/res/res_sip_session.c 394643 
> 
> Diff: https://reviewboard.asterisk.org/r/2683/diff/
> 
> 
> Testing
> -------
> 
> Hand testing against chan_sip and verification that behavior does not change when replaced with chan_sip.
> 
> 
> Thanks,
> 
> opticron
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20130718/559290c5/attachment-0001.htm>