[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