[asterisk-dev] [Code Review] Modularized RTP stack support

[Joshua Colp]

> On 2009-03-31 14:43:01, Russell Bryant wrote:
> > /trunk/UPGRADE.txt, line 26
> > <http://reviewboard.digium.com/r/209/diff/3/?file=3857#file3857line26>
> >
> >     There should be a comma after "not".

Fixed.


> On 2009-03-31 14:43:01, Russell Bryant wrote:
> > /trunk/include/asterisk/rtp_engine.h, lines 366-390
> > <http://reviewboard.digium.com/r/209/diff/3/?file=3871#file3871line366>
> >
> >     After some thought and out of band discussion with you about it, I think it makes sense to try to turn this into an opaque type.
> >     
> >     I would be happy to help with making the necessary changes.

Sounds good, working on it.


- Joshua


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On 2009-03-30 14:45:07, Joshua Colp wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> http://reviewboard.digium.com/r/209/
> -----------------------------------------------------------
> 
> (Updated 2009-03-30 14:45:07)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> This patch provides a common API (known as the RTP engine API) so that RTP stacks can be easily plugged into Asterisk. Functionality wise this patch should be equal to the current capabilities of our in-core RTP stack. The API is documented in the rtp_engine.h header file and the in-core RTP stack has been broken out into a module called res_rtp_asterisk.
> 
> 
> Diffs
> -----
> 
>   /trunk/main/rtp_engine.c PRE-CREATION 
>   /trunk/channels/chan_skinny.c 185119 
>   /trunk/channels/chan_unistim.c 185119 
>   /trunk/configs/sip.conf.sample 185119 
>   /trunk/include/asterisk/rtp.h 185119 
>   /trunk/include/asterisk/rtp_engine.h PRE-CREATION 
>   /trunk/include/asterisk/stun.h PRE-CREATION 
>   /trunk/main/Makefile 185119 
>   /trunk/main/asterisk.c 185119 
>   /trunk/main/loader.c 185119 
>   /trunk/main/rtp.c 185119 
>   /trunk/UPGRADE.txt 185119 
>   /trunk/apps/app_dial.c 185119 
>   /trunk/channels/chan_agent.c 185119 
>   /trunk/channels/chan_bridge.c 185119 
>   /trunk/channels/chan_gtalk.c 185119 
>   /trunk/channels/chan_h323.c 185119 
>   /trunk/channels/chan_jingle.c 185119 
>   /trunk/channels/chan_local.c 185119 
>   /trunk/channels/chan_mgcp.c 185119 
>   /trunk/channels/chan_sip.c 185119 
>   /trunk/main/stun.c PRE-CREATION 
>   /trunk/res/res_rtp_asterisk.c PRE-CREATION 
> 
> Diff: http://reviewboard.digium.com/r/209/diff
> 
> 
> Testing
> -------
> 
> I've tested using the most complex channel driver that uses the RTP stack, chan_sip. I've confirmed calls of various scenarios work but would like further testing with additional channel drivers that I am unable to test.
> 
> 
> Thanks,
> 
> Joshua
> 
>