[asterisk-dev] [Code Review] 2468: Pimp My SIP: SDES SRTP Support

[opticron]

> On April 25, 2013, 11:42 a.m., Joshua Colp wrote:
> > team/group/pimp_my_sip/include/asterisk/res_sip.h, lines 257-258
> > <https://reviewboard.asterisk.org/r/2468/diff/2/?file=36382#file36382line257>
> >
> >     I don't know if I like this none option. It's like optional SRTP. :P "I don't require it but if they offer it sure why not"

This is the way that chan_sip currently behaves for encryption=no as I understand the code (it's entirely possible that I read it wrong). Are you saying we want to drop the behavior I've specified for "no" and replace it with the behavior I've specified for "deny"?


> On April 25, 2013, 11:42 a.m., Joshua Colp wrote:
> > team/group/pimp_my_sip/res/res_sip_sdp_rtp.c, line 651
> > <https://reviewboard.asterisk.org/r/2468/diff/2/?file=36387#file36387line651>
> >
> >     This should have a configuration option for AVPF.

That's beyond the scope of this task, but I can go ahead and add it since it should be fairly trivial.


> On April 25, 2013, 11:42 a.m., Joshua Colp wrote:
> > team/group/pimp_my_sip/res/res_sip_session.c, lines 944-947
> > <https://reviewboard.asterisk.org/r/2468/diff/2/?file=36388#file36388line944>
> >
> >     This is ungood. It forces a dependency on having the SRTP module loaded, it also needlessly allocates memory if encryption has been disabled.

Fixed.


- opticron


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On April 24, 2013, 3:02 p.m., opticron wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2468/
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> 
> (Updated April 24, 2013, 3:02 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-21416
>     https://issues.asterisk.org/jira/browse/ASTERISK-21416
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> Add support for SDES SRTP in chan_gulp/res_sip.  Available options for media encryption for a given endpoint are deny (no encryption allowed), no (none offered by default, but silent upgrade to SDES allowed), and sdes.  This also supports mid-call rekeying as far as I could test it.  Much of the code necessary for this functionality was factored out of chan_sip or pulled from channels/sip/.
> 
> 
> Diffs
> -----
> 
>   team/group/pimp_my_sip/channels/sip/include/sip.h 386429 
>   team/group/pimp_my_sip/channels/sip/include/sdp_crypto.h 386429 
>   team/group/pimp_my_sip/channels/chan_sip.c 386429 
>   team/group/pimp_my_sip/channels/sip/include/srtp.h 386429 
>   team/group/pimp_my_sip/channels/sip/sdp_crypto.c 386429 
>   team/group/pimp_my_sip/channels/sip/srtp.c 386429 
>   team/group/pimp_my_sip/configs/res_sip.conf.sample 386430 
>   team/group/pimp_my_sip/include/asterisk/res_sip.h 386429 
>   team/group/pimp_my_sip/include/asterisk/res_sip_session.h 386429 
>   team/group/pimp_my_sip/include/asterisk/sdp_srtp.h PRE-CREATION 
>   team/group/pimp_my_sip/main/sdp_srtp.c PRE-CREATION 
>   team/group/pimp_my_sip/res/res_sip/sip_configuration.c 386429 
>   team/group/pimp_my_sip/res/res_sip_sdp_rtp.c 386429 
>   team/group/pimp_my_sip/res/res_sip_session.c 386429 
> 
> Diff: https://reviewboard.asterisk.org/r/2468/diff/
> 
> 
> Testing
> -------
> 
> Hand testing with several SRTP-capable endpoints and mid-call rekeying tested with minor tweaks to an otherwise unmodified chan_sip.
> 
> 
> Thanks,
> 
> opticron
> 
>

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