> On Dec. 19, 2013, 3:10 p.m., opticron wrote:
> > Ship It!
> 
> Matt Jordan wrote:
>     Well, hold on. Don't Ship It! just yet. :-)
>     
>     The fact that this isn't called in Asterisk 1.8 doesn't mean it shouldn't 
> be called - it just means someone, at some point in time, "fixed the glitch". 
> If we aren't calling this function, then we aren't able to do native UDPTL 
> bridges between fax sessions. While direct media and fax is dangerous (for 
> reasons explained on the -dev list awhile back), that doesn't mean that two 
> sessions that have negotiated the same fax parameters couldn't pass UDPTL 
> directly between each other without passing through the Asterisk core. Fax 
> pass through, as it were.
>     
>     I'd like to at least understand why this is no longer called before we 
> remove it. That means going back to 1.4/1.6 and determining why the code was 
> changed.
>     
>

It was never changed. The code was part of the contribution we received for 
UDPTL support (the RTP code was used as a base) but was never completed/used.


- Joshua


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3079/#review10461
-----------------------------------------------------------


On Dec. 19, 2013, 12:21 a.m., rmudgett wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3079/
> -----------------------------------------------------------
> 
> (Updated Dec. 19, 2013, 12:21 a.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> Removing dead code starting with ast_udptl_bridge() eliminated the code in 
> this patch.
> 
> Note: This code has actually been dead since at least Asterisk v1.8.
> 
> 
> Diffs
> -----
> 
>   /branches/12/main/udptl.c 404291 
>   /branches/12/include/asterisk/udptl.h 404291 
>   /branches/12/channels/chan_sip.c 404291 
>   /branches/12/addons/chan_ooh323.c 404291 
> 
> Diff: https://reviewboard.asterisk.org/r/3079/diff/
> 
> 
> Testing
> -------
> 
> It still compiles because nothing used what was deleted. :)
> 
> 
> Thanks,
> 
> rmudgett
> 
>

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Reply via email to