[asterisk-dev] [Code Review] 3337: Code for DTLS retransmission

[Matt Jordan]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3337/#review11706
-----------------------------------------------------------


(1) Don't forget to close out the findings once they're fixed!
(2) If you can, do file an issue in the ASTERISK project (if there isn't one already) related to this issue. When release summaries/announcements are made, it looks for the issue in the tracker. That helps people know what issues have been resolved.


http://svn.asterisk.org/svn/asterisk/branches/11/res/res_rtp_asterisk.c
<https://reviewboard.asterisk.org/r/3337/#comment21482>

    I think I was wrong in the order of operations here in my original finding (sorry). Since rtp is a property of instance, de'refing instance here first before unlocking it risks having the instance object being destroyed, which would also potentially destroy rtp. This should be:
    
    ast_mutex_unlock(&rtp->dtls_timer_lock);
    ao2_ref(instance, -1);



http://svn.asterisk.org/svn/asterisk/branches/11/res/res_rtp_asterisk.c
<https://reviewboard.asterisk.org/r/3337/#comment21483>

    Checking for rtp being a valid pointer is no longer necessary here, since you bail if rtp is NULL previously.


- Matt Jordan


On April 18, 2014, 10:14 a.m., Nitesh Bansal wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3337/
> -----------------------------------------------------------
> 
> (Updated April 18, 2014, 10:14 a.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> This patch adds the code to do the DTLS retransmissions in Asterisk.
> 
> 
> Diffs
> -----
> 
>   http://svn.asterisk.org/svn/asterisk/branches/11/res/res_rtp_asterisk.c 412566 
> 
> Diff: https://reviewboard.asterisk.org/r/3337/diff/
> 
> 
> Testing
> -------
> 
> I tested this with a basic SIPP script, which fakes a DTLS INVITE.
> Asterisk thinks that it is a DTLS call and inititates the DTLS handshake. SIPP doesn't respond to DTLS handshake, which causes the DTLS timeout and DTLS retransmission takes place.
> 
> 
> Thanks,
> 
> Nitesh Bansal
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140419/390238f3/attachment-0001.html>