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

[Joshua Colp]

> On 2009-03-31 14:49:54, Russell Bryant wrote:
> > /trunk/channels/chan_gtalk.c, line 1027
> > <http://reviewboard.digium.com/r/209/diff/3/?file=3861#file3861line1027>
> >
> >     Every usage of this API call passes &rtp_instance->codecs as the first argument, and the same rtp_instance as the second argument.  Do you forsee a use for this API call where this is not the case?  If not, you could simplify this by removing the first API call.  It would help a bit in making ast_rtp_instance opaque, too.

chan_sip actually uses the codecs API calls without an RTP instance. Only after it has figured everything out do they get copied into the RTP instance.


- Joshua


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On 2009-03-30 14:45:07, Joshua Colp wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> http://reviewboard.digium.com/r/209/
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> 
> (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
> 
>