[asterisk-dev] [Code Review] SRTP support for Asterisk

Terry Wilson twilson at digium.com
Tue May 4 20:03:19 CDT 2010



> On 2010-05-04 19:10:42, Russell Bryant wrote:
> > /trunk/CHANGES, line 62
> > <https://reviewboard.asterisk.org/r/191/diff/3/?file=9800#file9800line62>
> >
> >     SRTP only gets 3 words in CHANGES?  I think it deserves more than that.  :-)

It has more in there, it is just under the dialplan functions section. I went ahead and added some more info.


> On 2010-05-04 19:10:42, Russell Bryant wrote:
> > /trunk/channels/chan_iax2.c, lines 13640-13642
> > <https://reviewboard.asterisk.org/r/191/diff/3/?file=9802#file9802line13640>
> >
> >     sizeof(buf) is not valid here.  It should be 'buflen'

fixed


> On 2010-05-04 19:10:42, Russell Bryant wrote:
> > /trunk/channels/sip/sdp_crypto.c, lines 56-58
> > <https://reviewboard.asterisk.org/r/191/diff/3/?file=9808#file9808line56>
> >
> >     ast_calloc generates an error for you

fixed


- Terry


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/191/#review1964
-----------------------------------------------------------


On 2010-04-28 21:01:02, Terry Wilson wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/191/
> -----------------------------------------------------------
> 
> (Updated 2010-04-28 21:01:02)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> SRTP support for Asterisk using Sdescriptions. This has been sitting around for a while, so I figured that it should at least get some review.  Full description of setup at http://lists.digium.com/pipermail/asterisk-dev/2009-January/036029.html
> 
> 
> This addresses bug 5413.
>     https://issues.asterisk.org/view.php?id=5413
> 
> 
> Diffs
> -----
> 
>   /trunk/CHANGES 259665 
>   /trunk/build_tools/menuselect-deps.in 259665 
>   /trunk/channels/chan_iax2.c 259665 
>   /trunk/channels/chan_sip.c 259665 
>   /trunk/channels/sip/dialplan_functions.c 259665 
>   /trunk/channels/sip/include/sdp_crypto.h PRE-CREATION 
>   /trunk/channels/sip/include/sip.h 259665 
>   /trunk/channels/sip/include/srtp.h PRE-CREATION 
>   /trunk/channels/sip/sdp_crypto.c PRE-CREATION 
>   /trunk/channels/sip/srtp.c PRE-CREATION 
>   /trunk/configure UNKNOWN 
>   /trunk/configure.ac 259665 
>   /trunk/funcs/func_channel.c 259665 
>   /trunk/include/asterisk/autoconfig.h.in 259665 
>   /trunk/include/asterisk/frame.h 259665 
>   /trunk/include/asterisk/global_datastores.h 259665 
>   /trunk/include/asterisk/res_srtp.h PRE-CREATION 
>   /trunk/include/asterisk/rtp_engine.h 259665 
>   /trunk/main/asterisk.exports.in 259665 
>   /trunk/main/channel.c 259665 
>   /trunk/main/global_datastores.c 259665 
>   /trunk/main/rtp_engine.c 259665 
>   /trunk/makeopts.in 259665 
>   /trunk/res/res_rtp_asterisk.c 259665 
>   /trunk/res/res_srtp.c PRE-CREATION 
>   /trunk/res/res_srtp.exports.in PRE-CREATION 
> 
> Diff: https://reviewboard.asterisk.org/r/191/diff
> 
> 
> Testing
> -------
> 
> 
> Thanks,
> 
> Terry
> 
>




More information about the asterisk-dev mailing list