[asterisk-dev] [Code Review] 3275: res_rtp_asterisk: Fix the one way audio problems when resuming held calls with ICE

[Jonathan Rose]

> On Feb. 27, 2014, 2:14 p.m., Matt Jordan wrote:
> > /branches/11/res/res_rtp_asterisk.c, line 565
> > <https://reviewboard.asterisk.org/r/3275/diff/1/?file=54893#file54893line565>
> >
> >     You're actually leaking left_candidate here as well. The original ao2_link of left_candidate is one reference; the result of ao2_find is another. Make sure you drop the ref of left_candidate here as well.

Oi, thanks for pointing these out.  This also helped me spot another candidate leak on iteration in ast_rtp_ice_start.


- Jonathan


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On Feb. 27, 2014, 2:02 p.m., Jonathan Rose wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3275/
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> 
> (Updated Feb. 27, 2014, 2:02 p.m.)
> 
> 
> Review request for Asterisk Developers, Joshua Colp, Kevin Harwell, and Matt Jordan.
> 
> 
> Bugs: ASTERISK-22911
>     https://issues.asterisk.org/jira/browse/ASTERISK-22911
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> This patch provides a fix for the hold problem by doing the following:
> 
> Once an ICE session is marked as started, we start adding any new remote candidates into a separate list until we get another attempt to start the ICE session.
> Once a call to start the ice session is made, instead of immediately quitting if the session is already started, we check for a difference in the two candidates lists.  If the lists are identical, we wipe out the new list and keep the old one and just quit then going on with the current ICE session. If the lists are changed, we toss the old list and adopt the new one and restart the ICE session.
> 
> 
> Diffs
> -----
> 
>   /branches/11/res/res_rtp_asterisk.c 409132 
> 
> Diff: https://reviewboard.asterisk.org/r/3275/diff/
> 
> 
> Testing
> -------
> 
> SIPML client to Asterisk to Desk Phone
> SIPML calls desk phone
> audio test, got two way audio
> SIPML holds call
> SIPML resumes call
> audio test, got two way audio (previously this would cause one way audio from the SIPML client to the desk phone)
> 
> 
> Thanks,
> 
> Jonathan Rose
> 
>

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