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

[David Vossel]

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I'm in the process of making a full pass at this review.  I'm going to just make updates as I go along. This is everything I found to comment on up until sdp_crypto.c.


/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/191/#comment4620>

    The %30u used to get the port changed to a %30d.  I'd think we'd want this to continue to fail if someone gives us a negative port number, unless that actually means something in this case.  



/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/191/#comment4622>

    review https://reviewboard.asterisk.org/r/693 is changing how this is calculated.  You already know this though since you reviewed it, just thought I'd leave the comment as a reminder that this may change and need updating.



/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/191/#comment4623>

    I know the structure of this statement was here before your code, but it seems odd to me that a check to forbid direct RTP is dependent on the SIP_DIRECT_MEDIA flags not being set to ever even be executed.
    
    It seems this last "else if" statement should be a standalone if statement before checking the DIRECT_MEDIA flags at all.


- David


On 2010-06-01 21:57:09, Terry Wilson wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/191/
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> 
> (Updated 2010-06-01 21:57:09)
> 
> 
> 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 266876 
>   /trunk/build_tools/menuselect-deps.in 266876 
>   /trunk/channels/chan_iax2.c 266876 
>   /trunk/channels/chan_sip.c 266876 
>   /trunk/channels/sip/dialplan_functions.c 266876 
>   /trunk/channels/sip/include/sdp_crypto.h PRE-CREATION 
>   /trunk/channels/sip/include/sip.h 266876 
>   /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.ac 266876 
>   /trunk/funcs/func_channel.c 266876 
>   /trunk/include/asterisk/autoconfig.h.in 266876 
>   /trunk/include/asterisk/frame.h 266876 
>   /trunk/include/asterisk/global_datastores.h 266876 
>   /trunk/include/asterisk/res_srtp.h PRE-CREATION 
>   /trunk/include/asterisk/rtp_engine.h 266876 
>   /trunk/main/asterisk.exports.in 266876 
>   /trunk/main/channel.c 266876 
>   /trunk/main/global_datastores.c 266876 
>   /trunk/main/rtp_engine.c 266876 
>   /trunk/makeopts.in 266876 
>   /trunk/res/res_rtp_asterisk.c 266876 
>   /trunk/res/res_srtp.c PRE-CREATION 
>   /trunk/res/res_srtp.exports.in PRE-CREATION 
> 
> Diff: https://reviewboard.asterisk.org/r/191/diff
> 
> 
> Testing
> -------
> 
> 4 external tests written covering:
> Running with res_srtp noloaded to emulate a user not having libsrtp installed (to make sure we don't accidentally rely on SRTP support)
> Making a call with a user with encrypted=yes when libsrtp support is not enabled fails
> Making a call with encrypted=yes when libsrtp present results in an encrypted call (which also tests the CHANNEL(secure_media) function
> Using CHANNEL(secure_bridge_media) results in the outgoing call attempting to use encryption
> 
> In addition, I have tested a Polycom VVX-1500 to ensure that video + audio SRTP works.
> 
> 
> Thanks,
> 
> Terry
> 
>